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.

Audio transcoder mode switching depending on the input signal characteristics

Abstract: The invention relates to a method and apparatus for controlling the transition of a bypass capable codec between operative modes, based on a certain characteristic of the audio data signal processed by the codec. The apparatus relies on a control signal to determine when the codec will switch from one mode to another. This control signal reflects a characteristic of the audio data signal received at the apparatus, such as the type of speech activity or the format of the audio data signal. When in the active (non-bypass) mode, the apparatus relies on an additional control signal to switch to the inactive (bypass) mode. This additional control signal is received from a control unit at a remote codec that indicates that the remote codec is also bypass capable, hence the decoder at the first codec and the encoder at the remote codec can switch to the inactive mode to pass between them the compressed data frames.

The development of codec plug-in in OpenH323

Abstract: Codec plug-in in OpenH323 is used in media communication basing H323 protocol to make codec development easy and independent of OpenH323 frame structure. This paper introduces the OpenH323 fame structure and the details of codec plug-in. And a program for application and evaluation of the codec plug-in is given at last.

Audio Coding Standards, (Proprietary) Audio Compression Algorithms, and Broadcasting/Speech/Data Communication Codecs: Overview of Adopted Filter Banks

Abstract: In general, audio coding or audio compression algorithms are used to obtain compact digital representation of high-quality audio signals for their efficient transmission and storage. The central objective in audio coding is to represent the signal with a minimum number of bits while achieving its transparent reproduction. Besides speech coding schemes based on linear prediction methods especially tailored for efficient speech compression, the developed perceptual transform-based audio coding schemes gained a greater attention, particularly for applications in consumer electronics. Typically, any transform-based audio coding scheme utilizes a near-perfect quadrature mirror filter (QMF) and/or perfect reconstruction cosine-modulated filter bank to obtain a block-wise representation of the audio signal in the frequency domain. Perceptual transform-based audio coding schemes developed up to now are briefly reviewed including the family of ISO/IEC MPEG audio coding standards, proprietary audio compression algorithms, broadcasting/speech/data communication codecs, as well as open-free, patent royalty-free audio/speech codecs. The discussion is concentrated especially on adopted near-perfect QMF and perfect reconstruction cosine-modulated filter banks, processing methods, and specified transform block sizes.

Variable-rate video codec using frame adaptive finite-state vector quantization

Abstract: A new variable-rate codec for the compression of image sequences is presented. This codec is based on a two-dimensional finite-state vector quantization (2-D FSVQ), and a frame adaptive technique using codebook and address map replenishment. The results show that the codec achieves a good picture quality at low bit rate. >

Perceptual speech coding

Abstract: Investigates the applicability of perceptual coding to speech coding. The authors compare the performance of different filter banks on the quality of the reconstructed speech signal.