摘要:
Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.
摘要:
Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality.
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:
A method, medium, and system encoding and/or decoding an audio signal by extracting stereo parameters from an input signal, encoding the stereo parameters, and performing down-mixing on the input signal to a down-mixed signal, splitting the down-mixed signal into a low band signal and a high band signal, determining whether to encode the low band signal in a time domain or a frequency domain, if the low band signal is determined to be encoded in the time domain, encoding the low band signal in the time domain, if the low band signal is determined to be encoded in the frequency domain, generating an encoded bitplane by converting the low band signal from the time domain to the frequency domain by using a first conversion method and performing quantization and context-dependent encoding on the low band signal converted to the frequency domain by using the first conversion method, converting each of the low band signal and the high band signal from the time domain to the frequency domain or a time/frequency domain by using a second conversion method, generating and encoding bandwidth extension information that represents a characteristic of the high band signal converted by the second conversion method by using the low band signal converted by the second conversion method, and outputting the encoded stereo parameters, the encoded bitplane, and the encoded bandwidth extension information a result of encoding the input signal. Accordingly, high frequency components and stereo components may be efficiently encoded and decoded at a potential restricted bit rate, thereby improving the quality of an audio signal.
摘要:
Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.
摘要:
A method and apparatus to decode audio data constructed with a plurality of layers. An error concealment method of process a decoded bitstream selects one of a frequency domain and a time domain in order to conceal the errors, detects a position where the errors exist in a frame when the error concealment method in the frequency domain is selected, and conceals the errors only in a segment after the detected position.
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 multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.
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.
摘要:
An adaptive encoding method includes splitting an input signal into a low-frequency band signal and a high-frequency band signal; performing forward adaptive linear prediction on the low-frequency band signal and thus filtering the low-frequency band signal; selectively performing backward adaptive linear prediction or long-term prediction on the filtered low-frequency band signal according to the analysis result of the low-frequency band signal; transforming the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, into a signal in a frequency domain and quantizing the signal; and encoding the high-frequency band signal using the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, or the quantized signal. Therefore, compression efficiency of both speech and music signals can be enhanced, and a robust compression method can be provided for various audio contents at a low bit rate.