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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26 citations
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11 Apr 1988
TL;DR: Source coding and channel coding are embedded in a video coding algorithm for low bitrates (64 kb/s) to obtain a complete encoder which is resilient against channel errors.
Abstract: Source coding and channel coding are embedded in a video coding algorithm for low bitrates (64 kb/s) to obtain a complete encoder which is resilient against channel errors. The bits retained for the channel coder are spent for synchronization words, error correction, and update of erroneous parts in the image. The performance of the complete codec is evaluated by simulations in which channels with different characteristics are modeled. >
26 citations
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18 Jan 2005TL;DR: In this article, the authors proposed a method for compensating transient effects in transform coding and decoding of a combined speech and audio in electronic devices by using a transform based time-frequency domain codec.
Abstract: The present invention provides a method for compensating transient effects in transform coding and decoding of a combined speech and audio in electronic devices by using a transform based time-frequency domain codec. The method can combine, e.g., a CELP (code excited linear prediction) type speech codec and a transform type audio codec. The invention describes a compensation method to handle the transient (e.g., from the CELP coding to the transform coding) in transform coding when the number of quantized transform coding coefficients is lower than in the output of the transform.
25 citations
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16 Sep 1996TL;DR: This work replaces the arithmetic coding portion of the Taubman codec with block coding, and compares the encode/decode speed of this new coder with MPEG, which is one order of magnitude faster than MPEG-1.
Abstract: Scalable video compression is becoming increasingly more important in diverse, heterogeneous networks of today. In a previous work, Taubman and Zakhor (see IEEE Transactions on Image Processing, vol.3, no.5, p.572-88, 1994) developed a scalable codec capable of generating bit rates from tens of kilo bits per second to several mega bits per second with fine granularity of the available bit rates. This codec is based on 3-D subband coding and multi-rate quantization of subband coefficients, followed by arithmetic coding. We replace the arithmetic coding portion of the Taubman codec with block coding, and compare the encode/decode speed of this new coder with MPEG. Unlike MPEG, this codec requires symmetric computational power at the decoder and encoder and as such is useful in software only, real time, interactive video applications. We have found the encoding speed of the new encoder to be one order of magnitude faster than MPEG-1, without significant loss in compression efficiency.
25 citations
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21 Apr 1997TL;DR: A new approach to shape the coding noise in speech and audio coders, called spectral amplitude warping (SAW), consists essentially of a pre- and post-processing which apply a nonlinear transformation to the signal short-term spectrum prior to, and after, encoding.
Abstract: In this paper, we present a new approach to shape the coding noise in speech and audio coders. The approach, called spectral amplitude warping (SAW), consists essentially of a pre- and post-processing which apply a nonlinear transformation to the signal short-term spectrum prior to, and after, encoding. Since it is possible to view SAW as a separate entity from the coder, the noise shaping capability of an existing coder can be improved without modifying the coder itself. Using SAW as a pre- and post-process to the G.722 wideband speech coding standard, it was found in an informal listening test that the quality of the 64 kb/s operating mode can be achieved at only 48 kb/s. The price to be paid is an additional delay.
25 citations