Abstract:
A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal.
Abstract:
The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.
Abstract:
The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.
Abstract:
In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S301 the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S302 the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S303 codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.
Abstract:
The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.
Abstract:
The present invention intends to render quantization noise virtually imperceptible for a user and to prevent reduction in frequency resolution and reduction in encoding efficiency. A signal encoding apparatus includes: a quantization unit for quantizing an input signal based on a plurality of quantization methods; a dequantization unit for obtaining decoded signals by performing the dequantizing process; an error signal calculation unit for calculating a plurality of error signals between the decoded signals and the input signal; a weighting calculation unit for calculating, for each subblock, a weight related to degree concerning whether or not quantization noise corresponding to error signal is virtually imperceptible for a user; a quantization method selection unit for selecting a given quantization method from among the plurality of quantization methods, when a plurality of weighted error signals, obtained by assigning a weight of each subblock to an error signal of the subblock, are generated, based on the of weighted error signals; and an output unit for outputting the input signal quantized based on the given quantization method as an output signal.
Abstract:
A method and apparatus for high efficiency encoding of digital data whereby input digital data may be encoded by so-called high efficiency encoding. The high efficiency device includes a signal component encoding circuit including in turn a bandwidth storage circuit 62 for holding the information on the playback band of a previous block, that is the information on the bandwidth of bit allocation performed in the previous block, or the value of the number of the playback encoding units, and a control circuit 63 for deciding the number of playback encoding units of the current block based upon the playback band information, that is the value of the number of the playback encoding units, in the previous block, held by the bandwidth storage circuit 62. Since the stable playback bandwidth is maintained, and there is no risk of the playback band being frequently changed from one block to another, the harsh sounding noise otherwise produced by the appearance and disappearance of high-range side signals is not produced, so that deterioration in the perceived sound quality may be suppressed to a minimum with an insufficient bit rate.
Abstract:
A method and an apparatus for encoding an audio signal. The apparatus includes a converter converting the audio signal into a spectral representation; a scale parameter calculator calculating scale parameters; a spectral processor processing the spectral representation using the scale parameters; and a scale parameter encoder generating an encoded representation of the scale parameters. The scale parameter calculator is calculates an amplitude-related measure for each band to obtain a set of amplitude-related measures. A pre-emphasis operation is performed to the amplitude-related measures, so that low frequency amplitudes are emphasized with respect to high frequency amplitudes according to a tilt value, or a pre-emphasis factor. The scale parameter calculator controls the tilt value, or the pre-emphasis factor, based on a harmonicity measure of the audio signal.
Abstract:
A method for encoding an input point on an n-dimensional sphere by encoding n-1 spherical coordinates of said input point. The method includes sequential scalar quantization of the n-1 spherical coordinates in order to obtain at most 2n-2 candidates at the end of the sequential scalar quantization of the n-1 coordinates, and subsequently selecting the best candidate which minimizes a distance between the input point and the at most 2n-2 candidates, and determining the separate quantization indices resulting from the sequential scalar quantization of the spherical coordinates of the best candidate and sequentially encoding the separate quantization indices of the best candidate. A corresponding decoding method, an encoding device and a decoding device are also provided.
Abstract:
An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.