摘要:
Encoding and/or decoding a wideband signal produces high frequency band spectra from low frequency band spectral information. Linear prediction filter coefficients are determined for the entire wideband spectrum of an input signal. An energy value in each of a plurality of sub-bands in the high frequency band is determined and encoded. The short-term correlation removed input signal is then down-sampled to form a low frequency band signal. At a decoder, the high frequency band signal is generated using the encoded low frequency band signal. The energy in each sub-band of the high frequency band is adjusted using the encoded energy value. Thus, the spectral envelope for the entire wideband signal is synthesized and decoded using linear predictive synthesis.
摘要:
A method and apparatus to determine an encoding mode of an audio signal, and a method and apparatus to encode an audio signal according to the encoding mode. In the encoding mode determination method, a mode determination threshold for the current frame that is subject to encoding mode determination is adaptively adjusted according to a long-term feature of the audio signal for a frame (the current frame) that is subject to encoding mode determination, thereby improving the hit rate of encoding mode determination and signal classification, suppressing frequent oscillation of an encoding mode in frame units, improving noise tolerance, and improving smoothness of a reconstructed audio signal.
摘要:
An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.
摘要:
Provided is a method and apparatus to multiplex bitstreams that are coded to have different frame lengths using asynchronous time alignment, in which, based on the length of each frame of a bitstream selected as a reference bitstream from among bitstreams coded to have different frame lengths by a plurality of coders, the remaining bitstreams except for the reference bitstream are divided and multiplexed.
摘要:
An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.
摘要:
An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.
摘要:
Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.
摘要:
A method and apparatus to determine an encoding mode of an audio signal, and a method and apparatus to encode an audio signal according to the encoding mode. In the encoding mode determination method, a mode determination threshold for the current frame that is subject to encoding mode determination is adaptively adjusted according to a long-term feature of the audio signal for a frame (the current frame) that is subject to encoding mode determination, thereby improving the hit rate of encoding mode determination and signal classification, suppressing frequent oscillation of an encoding mode in frame units, improving noise tolerance, and improving smoothness of a reconstructed audio signal.
摘要:
Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.
摘要:
Provided is a method and apparatus for multiplexing bitstreams that are coded to have different frame lengths using asynchronous time alignment, in which, based on the length of each frame of a bitstream selected as a reference bitstream from among bitstreams coded to have different frame lengths by a plurality of coders, the remaining bitstreams except for the reference bitstream are divided and multiplexed.