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We have devised a method for representing frequency spectral envelopes with warped frequency resolution based on sparse non-negative matrices aiming at its use for frequency domain audio coding. With optimally prepared matrices, we can selectively control the resolution of spectral envelopes and enhance the coding efficiency. We show that the devised method can enhance the subjective quality of the...
We have devised a direct and simple scheme for linear conversion of line spectrum pairs (LSP) with low computational complexity aiming at weighting or inverse weighting spectral envelopes for noise control in speech and audio coders. Using optimally prepared coefficients, we can perform the conversion directly in the LSP domain, which ensures low computational costs and also simplifies the check or...
In this paper, we present an adaptive linear prediction order scheme - a simple and effective method to facilitate variable rate coding. We have applied it to the Extended Adaptive Multi-Rate Wideband (AMR-WB+), a state-of-the-art generic speech and audio coder with fixed-length coding designed for wireless transmission. By introducing this scheme into the codec, bit rate can be reduced with a negligible...
This paper describes two low-complexity tools used for the new ITU-T recommendation G.711.0, the lossless coding of G.711 (A-law/μ-law logarithmic PCM) speech data. One is an algorithm for quantizing the PARCOR/reflection coefficients and the other is an estimation method for the optimal prediction order. Both tools are based on a criterion that minimizes the entropy of the prediction residual signals...
The ITU-T Recommendation G.711 is the benchmark standard for narrowband telephony. It has been successful for many decades because of its proven voice quality, ubiquity and utility. A new ITU-T recommendation, denoted G.711.0, has been recently established defining a lossless compression for G.711 packet payloads typically found in IP networks. This paper presents a brief overview of technologies...
Along with the increase of all-digital movie theaters, the overarching service category called other digital stuff (ODS) has been drawing more and more attention. The target contents of ODS include not only plays and musicals but also a wide variety of entertainment such as concerts and sports. All are the target of strong demands for real-time streaming The development of an ultra-realistic streaming...
This paper presents enhanced codec software of the MPEG-4 Audio Lossless Coding (ALS) international standard for compressing multichannel time-series signals. By combining efficient software implementation and a fast search algorithm, multichannel data are efficiently compressed with practical computational load and extracted from the bitstream without any losses. Experimental results for instrumental...
Enhanced encoding tools for MPEG-4 audio lossless coding (ALS) international standard were developed, with the goal of improving compression performance of time-series biomedical data. The multichannel coding (MCC) tool of this standard exploits interchannel redundancies to reduce the bit rate. To improve compression performance with the MCC, we have devised a multichannel linear prediction tool,...
This paper describes a new search algorithm that quickly finds interchannel relationships between a coding channel and a reference channel in the multichannel coding tool of the MPEG- 4 Audio Lossless Coding (ALS) international standard. The algorithm has tree structure and can reduce data size with significantly smaller computation load than that of the conventional one. The devised method is based...
MPEG-4 audio lossless coding (ALS) is a standard compression tool for audio signals based on time- domain prediction. In this paper, enhanced encoder algorithms and implementation examples of MPEG-4 ALS are described and performance-complexity tradeoffs are discussed. Compression performance can be enhanced at the cost of computational complexity, using integrated tools, such as hierarchical block...
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