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.

A narrowband low bit rate sinusoidal audio and speech coder

Abstract: Speech and audio coders use different coding strategies. As a result, broadband (22.05 kHz bandwidth) audio coders typically have a good quality for both audio and speech signals and a high bit rate, whilst narrowband (4 kHz bandwidth) speech coders have a low bit rate but a distorted quality for audio signals. Our aim is to develop a single narrowband low bit rate audio and speech coder that provides speech quality on par with current state-of-the-art speech coders and good audio quality. To this end, the coder uses a mix of typical speech and audio coding strategies. A prototype coder was implemented and compared to the GSM-EFR standard speech coder. For speech signals, the attained quality approaches that of the GSM-EFR coder, whilst for music it performs clearly better. These results raise the expectation that more transparent coders are feasible for future mobile telephony.

8 kbit/s～32 kbit/s Wideband Codec Bitstream Interoperable with G.729:ITU-T G.729.1

Abstract: The principle of embedded wideband speech and audio codec ITU-T G.729.1 was introduced.G.729.1 can operate at 12 different bit rates from 32 kbit/s down to 8 kbit/s with wideband quality starting at 14 kbit/s.This coder is a bitstream interoperable extension of ITU-T G.729 based on three embedded stages: narrowband cascaded CELP coding at 8 kbit/s and 12 kbit/s,time-domain bandwidth extension(TDBWE) at 14 kbit/s,and split-band MDCT coding at 16 kbit/s and above.

Audio bandwidth detection in the EVS codec

Abstract: Speech and audio codecs are usually designed such that they encode all the frequency bands of the input signal spectrum. If the higher bands do not contain any perceptually meaningful content, these codecs often do not work optimally as they assign part of the available bit budget to encode these bands. In this paper we describe a bandwidth detection algorithm that determines the effective audio bandwidth of the input signal. This information is used to set the codec to its optimal configuration and consequently increase the coding efficiency for band-limited signals by allocating bits to encode only the useful bandwidth. The presented algorithm has been used in the new codec for Enhanced Voice Services (EVS), recently standardized by 3GPP, but it can be employed in other codecs as well.

High quality multi-rate CELP speech coding for wireless ATM networks

Abstract: The paper presents a multi-rate CELP speech coder which meets the requirements of high perceptive quality, robustness to noisy environments and flexibility required by speech communications in the wireless ATM scenario. The codec, exploiting new robust algorithms for multilevel phonetic classification and efficient activity/inactivity speech coding models, presents an average bit rate of 4 kbit/s with an algorithmic delay of 15 ms and a perceptive quality better than that of the GSM full-rate 13 kbit/s codec and similar to that of the 8 kbit/s ITU-T 0.729 standard, both in the discontinuous transmission mode. In addition, the paper demonstrates that the multi-rate speech coder proposed is a promising coding approach for future wireless ATM-based networks in that it exploits the bit rate variability within talkspurts, thus guaranteeing, with the same average bit rate as an on-off speech coder, greater robustness to packet loss and therefore a better quality of service (QoS).

From Lossy to Lossless:A Framework of Audio Codec

Abstract: Lossless audio technology is a kind of important audio coding technology which is used to archive digital audio data and encode high quality audioLossless audio codec system with lossy codec is describedIt includes lossless extension through lossy coding residual and independent lossless codecChannel decorrelation,in-teger lifting wavelet,linear prediction,residuals handling and arithmetic entropy coding are adoptedIn very com-plex conditions,the compression performance is achieved to internation mainstream lossless audio coding technolgy