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 codec identification through payload sampling

Abstract: W0065 present a new method for audio codec identification that does not require decoding of coded audio data. The method utilizes randomness and chaotic characteristics of coded audio to build statistical models that represent encoding process associated with different codecs. The method is simple, as it does not assume knowledge on encoding structure of a codec. It is also fast, since it operates on a block of data, which is as small as a few kilobytes, selected randomly from the coded audio. Tests are performed to evaluate the effectiveness of the technique in identification of the codec used in encoding on both singly coded and transcoded audio samples

An experimental audio codec based on warped linear prediction of complex valued signals

Abstract: Bark-scale warped linear prediction (WLP) is a very potential core for a monophonic perceptual audio codec. In the current paper the WLP scheme is extended for processing complex valued signals (CWLP). Three different methods of converting a stereo signal to one complex valued signal are introduced. The philosophy behind the coding scheme is to integrate some aspects of modern wideband audio coding (e.g. perceptuality and stereo signal processing) into one computational element in order to find a more holistic and economic way of processing.

Data CODEC system for computer

Abstract: The present invention relates to a data CODEC system for computer. The data CODEC system for computer comprises a system control software, a multichannel audio/speech and multimedia data signal processor, and a multichannel audio/speech and multimedia data input-output unit. The system control software communicates multichannel audio/speech and multimedia data with the multichannel audio/speech and multimedia data signal processor according to control of various application programs. The multichannel audio/speech and multimedia data signal processor processes multichannel audio/speech and multimedia data. The multichannel audio/speech and multimedia data input-output means inputs/outputs multichannel audio/speech and multimedia data from/to an external system.

Multi-layer scalable LPC audio format

Abstract: This paper presents a scalable audio format, called "multi-layer scalable LPC audio format", that addresses similar functionalities of MPEG-4. The format offers different levels of data rate and audio quality, and answers to the most important requirements of transmission and storage purposes, such as channel error robustness, cell loss robustness, low delay, and playback control. It operates in four modes. The first mode is based on a modified version of the LD-CELP algorithm, in which each 6 samples are represented by one single byte. In order to improve the signal-to-noise ratio (SNR), additional enhancement layers are embedded in the bit stream to allow higher quality at higher bit rates. The resultant bit rates are integer-multiple of 10.67 kbps. The other three modes use QMF splitting to two, four and eight subbands. These modes allow efficient representation of wideband audio and speech signals, and offer extension layers of 5.33 and 2.66 kbps. A simple and efficient header structure is embedded in the bitstream to allow the decoding process even in channel error conditions and even when the bitstream has been down-scaled somewhere during the transmission but has not been acknowledged to the decoder. Comparison results are conducted with respect to MPEG and ITU standards.

Adaptive transfer function for video encoding and decoding

Abstract: A video encoding and decoding system that implements an adaptive transfer function method internally within the codec for signal representation. A focus dynamic range representing an effective dynamic range of the human visual system may be dynamically determined for each scene, sequence, frame, or region of input video. The video data may be cropped and quantized into the bit depth of the codec according to a transfer function for encoding within the codec. The transfer function may be the same as the transfer function of the input video data or may be a transfer function internal to the codec. The encoded video data may be decoded and expanded into the dynamic range of display(s). The adaptive transfer function method enables the codec to use fewer bits for the internal representation of the signal while still representing the entire dynamic range of the signal in output.