Abstract: A quantization parameter signalling mechanism for both SDR and HDR content in video coding is described using two approaches. The first approach is to send the user-defined QpC table directly in high level syntax. This leads to more flexible and efficient QP control for future codec development and video content coding. The second approach is to signal luma and chroma QPs independently. This approach eliminates the need for QpC tables and removes the dependency of chroma quantization parameter on luma QP.

Abstract: An audio signal coding apparatus includes a time-frequency transformer that outputs sub-band spectra from an input signal; a sub-band energy quantizer; a tonality calculator that analyzes tonality of the sub-band spectra; a bit allocator that selects a second sub-band on which quantization is performed by a second quantizer on the basis of the analysis result of the tonality and quantized sub-band energy, and determines a first number of bits to be allocated to a first sub-band on which quantization is performed by a first quantizer; the first quantizer that performs first coding using the first number of bits; the second quantizer that performs coding using a second coding method; and a multiplexer.

Abstract: Disclosed is an encoding device, wherein the energy information of a given layer is efficiently encoded using a scalable encoding method in which the band to be encoded is selected in each layer, and the quality of decoded signals can be enhanced. An encoding device (101) is equipped with: a second layer encoding unit (205) which generates a second layer encoded information included in which is the first band information of said band; a second layer decoding unit (206) which generates a first decoding signal by using the second layer encoded information; an adding unit (207) which generates a second input signal by using the first decoding signal; and a third layer encoding unit (208) which generates a third layer encoded information included in which is a second band information obtained by selecting a second band to be quantized in the second input signal, and a corrected gain (energy information).

Abstract: A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.

Abstract: For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.

Abstract: An audio encoding method previously estimates better initial iterative values of global-gain and scalefactor for avoiding heavy calculation. The estimating process of the encoding method includes calculating the bit allocation of one frequency sample based on a sampling rate, a bit rate, and the number of audio channels according to an input frame, and the psychoacoustic model, searching one frequency sample having the greatest sample energy in each of a plurality of scalefactor bands, quantizing the frequency sample to comply with the bit allocation and to generate a corresponding scalefactor, searching a maximum scalefactor of all scalefactor bands corresponding to the input frame, and setting initial values of scalefactors and an initial value of global-gain for the quantization iterative loop process according to the corresponding scalefactor and the maximum scalefactor.

Abstract: Provided are a method and apparatus for coding and decoding an amplitude of partials, in which a step phenomenon can be prevented in the result of coding the amplitude of continuation partial partials in a parametric codec, thereby improving reproduced sound quality. The method of coding the amplitude of partials includes obtaining an inversely quantized amplitude of partials of a previous frame, determining a quantization level based on a function for the inversely quantized amplitude of the partials of the previous frame, and quantizing an amplitude of partials of a current frame based on the determined quantization level.

Abstract: A method, apparatus, and system for encoding or decoding a broadband voice signal are provided. The method includes extracting a linear prediction coefficient (LPC) from the broadband voice signal; outputting a linear prediction (LP) residual signal; pitch-searching a spectrum of the LP residual signal; extracting spectral magnitudes and phases of the LP residual signal, which correspond to a damping factor; obtaining, from among the extracted spectral magnitudes and phases, a first spectral magnitude and a first phase at which a power value of the LP residual signal is minimized; quantizing the first spectral magnitude and the first phase; and decoding the broadband voice signal. The apparatus includes a linear prediction coefficient (LPC) analyzer; an LPC inverse filter; a pitch searching unit; a sinusoidal analyzer; and a phase and spectral magnitude quantizer. The system includes a broadband voice encoding apparatus and a broadband voice decoding apparatus.