摘要:
A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.
摘要:
A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.
摘要:
An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.
摘要:
An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.
摘要:
An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.
摘要:
Disclosed is an apparatus for encoding and decoding a multi-channel audio signal. The apparatus for encoding the multi-channel audio signal groups channels of a multi-channel audio signal, eliminates redundant information between channels using a mixing matrix including phase information, converts a frequency of the signal, and encodes the signal.
摘要:
An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.
摘要:
A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.
摘要:
Disclosed is an apparatus for encoding and decoding a multi-channel audio signal. The apparatus for encoding the multi-channel audio signal groups channels of a multi-channel audio signal, eliminates redundant information between channels using a mixing matrix including phase information, converts a frequency of the signal, and encodes the signal.
摘要:
An apparatus and method for encoding/decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit/quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits.