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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Journal Article•
TL;DR: The interface design of 24 bit stereo audio Codec WM8731 based on FPGA, including program registers and audio digital interface is introduced, which can be controlled though registers only, ignoring its complicated clock timing requirements.
Abstract: 24 bit stereo audio Codec WM8731 has been widely applied in audio products because of its high quality and low power consumption.This paper introduces its interface design based on FPGA,including program registers and audio digital interface.Thus it can be controlled though registers only,ignoring its complicated clock timing requirements.The design is realized and verified in Max+Plus Ⅱ with VHDL and Verilog HDL,and results prove its practicality and convenience for designers.So designers can take use of it much easily,and gain some revelation as well when designing other interfaces.
1 citations
15 Nov 2002
TL;DR: TLS algorithms are applied to automatically extract the modeling parameters of a Exponential Sinusoidal Model (ESM) for audio signals to turn the SNR optimization criterion of these TLS algorithms into a perceptual modeling strategy and confirm that perceptual ESM achieves the same perceived quality as plain ESM while using substantially less components.
Abstract: Most parametric audio coders use a traditional sinusoidal model to represent the tonal parts of audio signals, together with dedicated models for the noise and transient-like parts of the audio signal. In this paper we apply Total Least Squares (TLS) algorithms to automatically extract the modeling parameters of a Exponential Sinusoidal Model (ESM) for audio signals. This sum of exponentially damped sinusoids is capable of accurately modeling most transient segments that are readily found in audio signals. In order to turn the SNR optimization criterion of these TLS algorithms into a perceptual modeling strategy we incorporate the psychoacoustic model of MPEG 1 Layer 1 into a subband TLS-ESM scheme. This allows us to model each subband in accordance with its perceptual relevance. Informal listening tests confirm that perceptual ESM achieves the same perceived quality as plain ESM while using substantially less components. The ESM model can serve as the parametric part of a hybrid audio coder in combination with traditional waveform coding techniques for the ESM residual signals.
1 citations
TL;DR: The results presents that the inter-view prediction can achieve higher efficiency in a multi-view video codec based on a high performance video codec such as HEVC.
Abstract: Multi-view video coding (MVC) is a video coding standard developed by MPEG and VCEG for multi-view video. It
showed average PSNR gain of 1.5dB compared with view-independent coding by H.264/AVC. However, because
resolutions of multi-view video are getting higher for more realistic 3D effect, high performance video codec is needed.
MVC adopted hierarchical B-picture structure and inter-view prediction as core techniques. The hierarchical B-picture
structure removes the temporal redundancy, and the inter-view prediction reduces the inter-view redundancy by
compensated prediction from the reconstructed neighboring views. Nevertheless, MVC has inherent limitation in coding
efficiency, because it is based on H.264/AVC. To overcome the limit, an enhanced video codec for multi-view video
based on Key Technology Area (KTA) is proposed. KTA is a high efficiency video codec by Video Coding Expert
Group (VCEG), and it was carried out for coding efficiency beyond H.264/AVC. The KTA software showed better
coding gain than H.264/AVC by using additional coding techniques. The techniques and the inter-view prediction are
implemented into the proposed codec, which showed high coding gain compared with the view-independent coding
result by KTA. The results presents that the inter-view prediction can achieve higher efficiency in a multi-view video
codec based on a high performance video codec such as HEVC.
1 citations
12 Mar 2015
TL;DR: The performance of H.264 Video codec is evaluated and analyzed for various advance options such as context adaptive binary arithmetic coding (CABAC), which can be up to 32% better when compared to other entropy compression methods such as Huffman Coding.
Abstract: This paper deals with the evaluation, analysis and algorithm of advance options of H.263 and H.264/AVC video codec. Analysis and evaluation of H.263 video codec is important because it enable us to determine output parameter, which is peak signal to noise ratio (PSNR) or quality of picture. In this paper, the performance of H.263 video codec is evaluated and analyzed for various advance options such as syntax based arithmetic coding (SBAC), advanced prediction mode, unrestricted motion vector mode etc. The H.264/AVC is an international video coding standard, jointly developed by groups from ISO/IEC and ITU-T, which aims at achieving improved compression performance. Various applications of H.264/AVC include conversational, storage, and streaming. For H.264 Video codec we proposed algorithm for advance options such as context adaptive binary arithmetic coding (CABAC). The improvement in video quality with the use of these advance options of H.263 is of the order of 0.8 dB or more. The compression efficiency of CABAC is 10–15% better as compared to context adaptive variable length coding (CAVLC) and it can be up to 32% better when compared to other entropy compression methods such as Huffman Coding.
1 citations
Patent•
13 Aug 1998
TL;DR: For a fixed, non-zero time interval (Tmin) as mentioned in this paper, the passage from a level of coding of rank m, where 0 ≤ m less than n-1 and n −1 corresponds to a minimum protection, to rank m+q, where m ≤ m −1 and q ≤ m−1 is positive, is prevented when the received signal quality correspond to a coding level of m + q. The loss of quality arising from a sub-optimal choice of coding level is compensated by preserving frames which would otherwise be lost
Abstract: For a fixed, non-zero time interval (Tmin) the passage from a level of coding of rank m, where 0 <= m less than n-1 and n-1 corresponds to a minimum protection, to rank m+q, where 0 less than m+q <= n-1 and q is positive, is prevented when the received signal quality corresponds to a coding level of m+q The loss of quality arising from a sub-optimal choice of coding level is compensated by preserving frames which would otherwise be lost
1 citations