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: All aspects of this standardization effort are outlined, starting with the history and motivation of the MPEG work item, describing all technical features of the final system, and further discussing listening test results and performance numbers which show the advantages of the new system over current state-of-the-art codecs.
Abstract: In early 2012 the ISO/IEC JTC1/SC29/WG11 (MPEG) finalized the new MPEG-D Unified Speech and Audio Coding standard The new codec brings together the previously separated worlds of general audio coding and speech coding It does so by integrating elements from audio coding and speech coding into a unified system The present publication outlines all aspects of this standardization effort, starting with the history and motivation of the MPEG work item, describing all technical features of the final system, and further discussing listening test results and performance numbers which show the advantages of the new system over current state-of-the-art codecs
42 citations
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09 Dec 1997TL;DR: In this article, a portable multimedia data input/output processor consisting of audio codec for compressing and decompressing audio data, video codec controller and multimedia processor for transmitting audio data to wireless communication controller and video data to video codec and to graphic processor.
Abstract: Potable multimedia terminal which is small and consumes low power, can process a large quantity of multimedia data such as video and audio data. Portable multimedia data input/output processor can be made smaller by using a pen as an input device and can also process a large quantity of multimedia data at a high speed by adopting a PCI bus as a local bus of a system. To retrieve, compress, and decompress multimedia data, main components of this portable multimedia data input/output processor are comprised of audio codec for compressing and decompressing audio data, video codec controller for compressing and decompressing video data, and multimedia processor for transmitting audio data to wireless communication controller and video data to video codec controller and to graphic processor. The method for retrieving multimedia data includes steps of receiving data, de-interleaving received data into audio, video, and graphic data, decompressing the data, and outputting the data to output device. The method for compressing data includes steps of inputting video data to video codec controller, compressing video and audio data at video codec controller and audio codec, interleaving the compressed data, and transmitting them to a remote system. The steps to decompress data are in reverse to the steps to compress data.
42 citations
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TL;DR: An objective evaluation of WebP is provided, by comparing it with the JPEG family algorithms, and it appears that the performance of the proposed codec is in line with that of the alternative methods, without achieving any major improvement and lacking several features.
Abstract: Research on multimedia data coding is allowing for better performance in terms of compression ratio, coding features, and robustness against transmission errors. While rate-distortion performance is being improved at a slower pace if compared to what we were used to up to a decade ago, remarkable advances are being made by adding complex features, such as fast adaptive transforms, lossy to lossless coding, compressed domain processing, etc. One of the latest codec which is expected to improve on the state of the art is the WebP algorithm released by Google. In this paper we provide an objective evaluation of WebP, by comparing it with the JPEG family algorithms. From the results it appears that the performance of the proposed codec is in line with that of the alternative methods, without achieving any major improvement and lacking several features.
42 citations
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Université de Sherbrooke1, Texas Instruments2, Nokia3, Motorola4, Ericsson5, Panasonic6, Huawei7, Orange S.A.8
TL;DR: ITU-T Embedded Variable Bit-Rate (EV-VBR) codec is presented, being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718, robust to significant rates of frame erasures or packet losses and several technologies are used to encode the MDCT coefficients for best performance both for speech and music.
Abstract: This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Sev-eral technologies are used to encode the MDCT coefficients for best performance both for speech and music. The codec performance is demonstrated with selected results from ITU-T Characterization test.
42 citations
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23 Apr 2009TL;DR: In this paper, a method to establish a full-duplex audio connection over an asynchronous Bluetooth link between an audio terminal and a wireless audio device exchanges supported service classes and codecs between the audio terminals and the wireless audio devices.
Abstract: A method to establish a full-duplex audio connection over an asynchronous Bluetooth link between an audio terminal and a wireless audio device exchanges supported service classes and codecs between the audio terminal and the wireless audio device, negotiates a service class and a codec that are common to the audio terminal and the wireless audio device, and establishes an asynchronous audio connection between the audio terminal and the wireless audio device using the common service class and the codec. The audio connection established can depend on the software application desiring the audio connection plus the available service classes and codecs at the audio terminal and wireless audio device. For non-internet protocol (non-IP) audio applications, an ACL using AVDTP may be selected; for IP audio applications, an ACL using BNEP may be selected. Both AVDTP and BNEP can use codecs that support wide bandwidth audio.
41 citations