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.

Joint DC coefficient band decoding and motion estimation in Wyner-Ziv video coding

Abstract: In contrast to traditional predictive coding, Wyner-Ziv video coding enables low-cost encoding architectures, in which the computationally expensive tasks for performing motion estimation are shifted to the decoder-side. In Wyner-Ziv video coding, side-information generation is a key aspect profoundly affecting the compression capacity of the system. This paper presents a novel technique which enables side-information refinement after DC coefficient band decoding in a transform-domain Wyner-Ziv video codec. The proposed side-information refinement approach performs overlapped block motion estimation and compensation, utilizing multi-hypothesis pixel-based prediction. The experimental results show that the presented Wyner-Ziv video codec incorporating the proposed technique yields significant and systematic compression gains of up to 23.22% with respect to the state-of-the-art DISCOVER codec.

Hybrid audio encoder and hybrid audio decoder

Abstract: Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method.

Fast time-frequency transform algorithms and their applications to real-time software implementation of AC-3 audio codec

Abstract: AC-3 audio coding technology is a kind of perceptual audio coder (PAC) developed by the Dolby Company. Up to 5 full-bandwidth channels and one subwoofer channel (cutoff at 120 Hz) are available in AC-3 to provide multi-channel, low bit rate, and high perceptual quality of audio. This explains why AC-3 has become the audio standard of many international standards. In this paper, we focus on the real-time software implementation issues of AC-3. Two fast algorithms of the time-frequency transform, one for memory economization and the other is for time domain subsampling, are presented. Meanwhile, we state the current performance status of our AC-3 decoder.

Method and system for multimedia conferencing

Abstract: A multipoint control unit (104) is provided which allows for dynamic codec selection According to one embodiment, the MCU (104) causes endpoints (102, 106) to renegotiate their codec selections if a most-commonly available codec is not being used, upon entry of new parties to a teleconference Alternatively, the codec renegotiation may be performed each time a user speaks, to optimize for maximum transmission quality or for minimizing transcoding

A progressive algorithm for perceptual coding of digital audio signals

Abstract: In this paper, a progressive audio coding algorithm is presented. Distinctive from existing transform-based perceptual audio coding schemes, which allow either constant bit rate or multiple fixed bit rates, our proposed algorithm achieves a fully embedded audio coding scheme whose bit rate can be controlled depending on the user's requirement. This feature makes it extremely useful in audio networking applications, such as audio on-demand and audio broadcasting over the Internet. In the proposed audio coding scheme, efficient successive quantization and bit layer coding are applied to the compact audio signal representation obtained from traditional subband decomposition. Our research focuses on the improved successive approximation quantisation (ISAQ), which performs bit allocation for quantization based on psychoacoustic analysis of the input audio signal. Thus the progressive audio coding is achieved, while perceptually transparent audio coding is also maintained to the allowance of bit rate constraint. The performance measurement of our presented algorithm is demonstrated.