A fast noise-scaling algorithm for uniform quantization in audio coding schemes
نویسندگان
چکیده
A new bit assignment algorithm is presented. Its goals are the simultaneous assignment on all subbands in a few steps of an iterative calculus, the use of memory to achieve a better speed of convergence and the consideration of a deformable error curve. The basis of the algorithm is discussed and also other considerations that are likely to arise in practice. Finally, an example of performance is given.
منابع مشابه
Coding Algorithm Based on Loss Compressing using Scalar Quantization Switching Technique and Logarithmic Companding
This paper proposes a novel coding algorithm based on loss compression using scalar quantization switching technique. The algorithm of switching is performed by the estimating input variance and further coding with Nonuniform Switched Scalar Compandor (NSSC). An accurate estimation of the input signal variance is needed when finding the best compressor function for a compandor implementation. I...
متن کاملVery low bit rate parametric audio coding
In this thesis, a parametric audio coding system for very low bit rates is presented. It is based on a generalized framework that combines different source models into a hybrid model and thereby permits flexible utilization of a broad range of source and perceptual models. The developed parametric audio coding system allows efficient coding of arbitrary audio signals at bit rates in the range o...
متن کاملQuantization Noise Power Injection In Subband Audio Coding Using Low Selectivity Filter Banks
The ISO-MPEG standards are widely used for high quality audio coding. They use perceptual models that require high frequency resolution. This implies two characteristics: they use high selectivity filter bank to decompose the audio signal, and the FFT frame length for the frequency masking threshold calculations is long. Both characteristics carried out a very high delay coding. In applications...
متن کاملLloyd-Max's Algorithm Implementation in Speech Coding Algorithm Based on Forward Adaptive Technique
In this paper a detail analysis of speech coding algorithm based on forward adaptive technique is carried out. We consider an algorithm that works on frame-by-frame basis, where a frame consists of a certain number of speech samples. Buffering frame-by-frame an estimation of the gain defined as squared root of the frame variance is enabled. The information about the gain (side information) and ...
متن کاملImproving perceptual coding of narrowband audio signals at low rates
This paper discusses perceptual coding of narrowband audio signals at low rates. In particular, it proposes a new error measure which shapes the noise inside the critical bands, a window switching criterion based on the temporal masking effect of the hearing system, a more accurate model of the simultaneous masking effect of the hearing system, perceptually-based bit allocation algorithms based...
متن کاملذخیره در منابع من
با ذخیره ی این منبع در منابع من، دسترسی به آن را برای استفاده های بعدی آسان تر کنید
عنوان ژورنال:
دوره شماره
صفحات -
تاریخ انتشار 1997