The present invention relates to image processing apparatus and method which make it possible to prevent a decrease in compression efficiency without increasing computational complexity An intra-TP motion prediction/compensation unit 75 performs motion prediction within a predetermined search range by taking predicted motion vector information generated by an intra-predicted motion vector generating unit 76 as the center of search, on the basis of an image to be intra-predicted from a screen rearrangement buffer 62, and reference images from a frame memory 72 An inter-TP motion prediction/compensation unit 78 performs motion prediction within a predetermined search range by taking predicted motion vector information generated by an inter-predicted motion vector generating unit 79 as the center of search, on the basis of an image to be inter-encoded from the screen rearrangement buffer 62, and reference images from the frame memory 72 The present invention can be applied to, for example, an image encoding apparatus that performs encoding in H264/AVC format

Image Processing Apparatus and Method

A method for processing an audio signal including classifying an input frame as either a speech frame or a generic audio frame, producing an encoded bitstream and a corresponding processed frame based on the input frame, producing an enhancement layer encoded bitstream based on a difference between the input frame and the processed frame, and multiplexing the enhancement layer encoded bitstream, a codeword, and either a speech encoded bitstream or a generic audio encoded bitstream into a combined bitstream based on whether the codeword indicates that the input frame is classified as a speech frame or as a generic audio frame, wherein the encoded bitstream is either a speech encoded bitstream or a generic audio encoded bitstream.

Embedded speech and audio coding using a switchable model core

RISC-V is an open-source instruction set-architecture, designed to support customized extensions and architectures. This paper presents an instruction-set extension to the RISC-V ISA, idealized for software-defined radio applications. The custom instructions perform complex-number arithmetic, tailored for complex or quadrature modulation and baseband processing, and can perform one complex multiply-accumulate per cycle. The proposed system architecture includes the processor core, a WISHBONE bus interconnection, IO and peripherals, and was targeted to an Altera Cyclone III FPGA, achieving 0.9 DMIPS/MHz without the use of any compiler optimizations.

Oolong: A Baseband processor extension to the RISC-V ISA

An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.

Audio decoding apparatus, audio coding apparatus, and system comprising the apparatuses

The embodiments of the present invention disclose an audio signal reconstruction method and device. The method includes the following steps that: compressed data corresponding to at least two audio signals are acquired; inverse quantization is performed on the compressed data corresponding to at least two audio signals so as to obtain measurement data corresponding to the at least two audio signals; a measurement matrix is obtained, sparse transformation coefficients corresponding to the at least two audio signals are jointly reconstructed according to the measurement data and measurement matrix corresponding to the at least two audio signals; and inverse sparse transformation is performed on the sparse transformation coefficients corresponding to the at least two audio signals, so that the at least two audio signals can be obtained. With the audio signal reconstruction method and device of the embodiments of the present invention adopted, the quality of the audio signals can be improved.

Audio signal reconstruction method and device

A gradable combined voice and music CODEC device in the technical domain of digital audio encoding is provided In the encoding unit in the present invention, a voice and music grading unit carries out gradation for audio signals, a graded voice encoding unit and a graded music residual encoding unit carry out encoding for graded voice signals, a graded music encoding unit and a graded voice residual encoding unit encoding unit carry out encoding for graded music signal, and provide output via a combined output module; the decoding unit comprises an encoding mode obtaining unit, an encoding parameter obtaining unit, a graded voice decoding unit, a graded music decoding unit, a combined output unit, and is designed to accomplish the process reverse to the process of the encoding unit The CODEC device provided in the present invention is applicable to all voice and music signals, and can reduce data loss in the encoding process, improve encoding quality; especially, the CODEC device can maintain high acoustic quality at higher code rates

Apparatus for encoding and decoding hierarchical voice and musical sound together

For next-generation audio applications, the dominant trends are much higher sample rate, larger word length and more audio channels for playback audio data. Traditional DSPs or embedded processors are inefficient for such kinds of applications because of their non-specific or limited computing capabilities as well as the on-chip memory architectures. In this paper, an embedded audio processor aiming at next-generation audio applications has been proposed. The audio specific instruction set architecture is based on the analysis of the requirements for next-generation audio processing. Besides, a novel tightly coupled audio memory has been proposed to support extremely high audio data throughputs and flexible audio data transfers with main memories. To evaluate the performance of the proposed audio processor, a set of benchmarks have been used based on the analysis of next-generation audio applications. The implementation and evaluation results lead to the conclusion that the proposed audio processor is of outstanding efficiency and cost-effectiveness for next-generation audio applications.

Next-generation consumer audio application specific embedded processor

Embedded algebraic vector quantization (EAVQ) is a fast and efficient lattice vector quantization (LVQ) scheme used in low-bitrate audio coding. However, a defect of EAVQ is the overload distortion which causes unpleasant noises in audio coding. To solve this problem, specific base codebook extension schemes should be carefully considered. In this letter, we present a novel EAVQ codebook extension scheme-split table extension (STE), which splits a vector into two smaller vectors: one in the base codebook and the other in the split table. The base codebook and the split table are designed according to the appearance probability of quantized vectors in audio segments. Experiments on encoding multiple audio and speech sequences show that, compared with the existed Voronoi extension scheme, STE greatly reduces computation complexity and storage requirement while achieving similar coding quality.

Split Table Extension: A Low Complexity LVQ Extension Scheme in Low Bitrate Audio Coding

A variable length split table (VLST) based VQ high-order code book extended CODEC system in CODEC domain is provided; the decoding system comprises an extended code judging module, a coding frame vector splitting module, a fundamental code book encoding module, an extension mode selecting module, a primary extension calculation module, a secondary extension calculation module, and a packet output module; the decoding system comprises a parameter obtaining module, an encoding mode judging module, a fundamental code book decoding module, a primary extension split decoding module, a secondary extension split decoding module, and a vector composition module. Code words that are not found in the fundamental code book are encoded in a VLST-based two-stage extension mode, wherein, fixed-length bit code splitting is employed for primary extension; variable-length bit code splitting is used for secondary extension. Such a two-stage extension scheme can ensure high quantization efficiency for shorter code words and reduce overhead bits in vector quantization of longer code words.

Jin Wang

Papers

Apparatus for encoding and decoding hierarchical voice and musical sound together

Next-generation consumer audio application specific embedded processor

Split Table Extension: A Low Complexity LVQ Extension Scheme in Low Bitrate Audio Coding

Expanding encoding and decoding system based on vector quantization high-order code book of variable splitting table