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.

System and method for providing an enhanced audio quality telecommunication session

Abstract: An improved telecommunication system is capable of supporting an enhanced audio transmission mode and a conventional PCM waveform encoding mode. The enhanced audio transmission mode is governed by an audio coding protocol, while the PCM mode is governed by a PCM protocol such as μ-law encoding. The telecommunication system performs an in-band signaling routine during a first communication session in accordance with the PCM protocol. The in-band signaling routine employs a form of robbed bit signaling to transmit information between the calling codec and the called codec. The signaling information is utilized to determine whether the called codec is compatible with the enhanced audio coding mode and, as necessary, to initiate the transition between the PCM mode and the audio coding mode. The audio coding mode transmits signals using a wider bandwidth than that used during the PCM mode. The use of a wider bandwidth results in a higher quality sound that better resembles person-to-person speech.

A VR-CELP codec implementation for CDMA mobile communications

Abstract: This paper focuses on the design, implementation and testing of a variable rate (VR) CELP codec aimed to be used in the testbed of one RACE-II project: CoDiT (code division testbed). The project has been conceived to demonstrate the potentiality of CDMA for the UMTS (universal mobile telecommunications system). Because of the flexibility permitted by CDMA to easily convey the information stream over a VR physical channel, the fixed-rate constraint has been removed from the speech coding algorithm design, in order to exploit the time-varying local character of speech. One major feature of the proposed algorithm is the possibility for the average rate to be either source-controlled or network-controlled. This is particularly appealing for cellular communications in order to cope with areas or cells with a high time-varying congestion. >

Joint speech codec parameter and channel decoding of parameter individual block codes (PIBC)

Abstract: In digital mobile speech transmission usually the most important (class la) bits provided by the speech coding scheme are protected by a CRC for error detection. As a consequence all parameters spanned by the class la bits have to be marked at the receiver either as reliable or as unreliable. In contrast to this somewhat coarse approach we propose the usage of what we call parameter individual block codes (PIBC) for the most important codec parameters. This allows joint speech codec parameter and PIBC decoding taking advantage of the error concealing properties of soft-bit speech decoding.

Video codec system with real-time complexity adaptation

Abstract: An encoder/decoder (codec) is configured to cause the coding/decoding algorithms used by the codec to dynamically adapt according to the available computational resources in response to actual complexity measurements performed at run-time. In a system in which multiple video codecs have to operate simultaneously in real-time, sharing the system's available resources, this invention provides a way to increase the number of simultaneous codecs that can co-exist. The principles of the invention are also applicable to an individual encoder and an individual decoder.