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.

Research on low bit rate speech coding at the Electrical Communication Laboratory, NTT

Abstract: Since the introduction of speech analysis—synthesis based on the maximum likelihood spectrum estimation—in 1966, we have been conducting research activities on low bit rate speech coding techniques, and their aplication to audio response and low bit rate digital speech transmission. Parcor analysis‐synthesis, demonstrated in 1969, was one of the most fundamental methods, and it has formed the basis of the present development of linear predictive coding. Recently, various kinds of techniques have been proposed to improve speech quality, such as interpolation and nonlinear quantization of parameters, spectral smoothing, etc. They have been applied in the hardware realization of a 4 CH multiplexed 2400 bps Vocoder. At present, the Parcor method is being applied to a wide range of speech coding from 1200 bps variable frame‐rate coding to high quality 16 kbps adaptive, predictive coding.

Audio Coding Based on Long Temporal Contexts

Abstract: We describe novel audio coding technique designed to be utilized at medium bit-rates. Unlike classical state-of-the-art audio coders that are based on short-term spectra, our approach uses relatively long temporal segments of audio signal in critical-band-sized sub-bands. We apply auto-regressive model to approximate Hilbert envelopes in frequency sub-bands. Residual signals (Hilbert carriers) are demodulated and thresholding functions are applied in spectral domain. The Hilbert envelopes and carriers are quantized and transmitted to the decoder. Our experiments focused on designing audio coder to provide broadcast radio-like quality audio around $10-20$kbps. Objective quality measures indicate comparable performance with the 3GPP-AMR speech codec standard for both speech and non-speech signals.

Perceptual Pre-weighting and Post-inverse weighting for Speech Coding

Abstract: We investigate the effect on voice quality of perceptual pre-weighting of the input speech to a codec, and post- inverse weighting the output of the codec. The G.726 adaptive differential pulse code modulation (ADPCM) codec and the AMR narrowband (AMR-NB) code excited linear prediction (CELP) codec are employed in our experiments. The weighting function used has the same form as that of the perceptual weighting function for the analysis-by-synthesis codebook search in AMR- NB. We observe a significant improvement in voice quality at rates of 16 and 24 kbps in the case of G.726 when perceptual weighting is used. When we use pre-weighting with the AMR codec, the unweighted squared error is used within the analysis- by-synthesis codebook search loop, and we find that the quality of the pre-weighted approach is comparable to the quality achieved by the standard AMR codec. The proposed pre-weighting method requires an additional bit-rate of 1.35 kbps to communicate the linear prediction (LP) coefficients of the original speech input to the decoder.

Daala: A Perceptually-Driven Still Picture Codec

Abstract: Daala is a new royalty-free video codec based on perceptually-driven coding techniques. We explore using its keyframe format for still picture coding and show how it has improved over the past year. We believe the technology used in Daala could be the basis of an excellent, royalty-free image format.