摘要:
A module capable of appropriately selecting a linear predictive coding (LPC)-based or a code excitation linear prediction (CELP)-based speech or audio encoder and a transform-based audio encoder according to a feature of an input signal is a module that performs as a bridge for overcoming a performance barrier between a conventional LPC-based encoder and an audio encoder. Also, an integral audio encoder that provides consistent audio quality regardless of a type of the input audio signal can be designed based on the module.
摘要:
Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.
摘要:
Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.
摘要:
Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.
摘要:
Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.
摘要:
Provided are a lossless audio coding/decoding apparatus and method. The lossless audio coding apparatus includes a first coder to directly code first symbols; a second coder module comprising a plurality of second coders to convert the first symbols into second symbols and to code the second symbols; a first selector to compare the performance of the first coder to the performance of the second coders and to output a coding mode in accordance with a comparison result; and a second selector to output a final bitstream by coding the first symbols in correspondence with the coding mode. According to the present invention, the performance of audio coding may be improved.
摘要:
An apparatus and method for deciding an adaptive noise level for bandwidth extension are provided. The apparatus includes a noise level decider for deciding a high-band noise level for bandwidth extension according to tonality of an input signal, a pitch frequency analyzer for detecting a pitch frequency of the input signal and analyzing correlation between the detected pitch frequency and a frequency channel, and a noise level controller for adaptively controlling the decided high-band noise level based on the analyzed correlation of the pitch frequency and the frequency channel.
摘要:
A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.
摘要:
The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.
摘要:
Disclosed are an exhaust gas treating apparatus and a treating method for a carbon dioxide capture process, in which harmful substances remaining in the exhaust gas discharged from the conventional flue-gas desulfurization process are additionally removed for efficient performance of the carbon dioxide capture process. According to the exhaust gas treating apparatus for a carbon dioxide capture process, it has the effects of minimizing the installation space of desulfurization equipment and reducing the process cost. In addition, by keeping the contaminants contained in the gas introduced in the carbon dioxide capture equipment below a proper level, absorption performance can be improved as degradation of the absorbent used in the carbon dioxide capture process is prevented. After all, it has an advantage of preventing the pollution by the exhaust gas discharged into the atmosphere.