摘要:
A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.
摘要:
Provided are a method and apparatus for encoding or decoding an audio signal or a speech signal. In the encoding method, encoding is performed by performing domain transformation on a received signal in units of frequency bands by applying a psychoacoustic model, encoding the transformation result with respect to predetermined one or more frequency bands by using a high temporal resolution coding tool, and then quantizing the encoding result. In the decoding method, decoding is performed by inversely quantizing signals obtained by encoding in units of frequency bands, decoding one or more signals from among the inversely quantized signals, which are allocated to one or more frequency bands which have a predetermined domain resolution, determined by applying the psychoacoustic model, that is greater than a predetermined value, according to a predetermined method, and then inversely transforming either the inversely quantized or the one or more decoded signals.
摘要:
A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.
摘要:
A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.
摘要:
A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.
摘要:
A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.
摘要:
Provided is a method and apparatus for encoding a bitstream, which was encoded by a predetermined method, in another method. By adaptively encoding a bitstream encoded by a predetermined method by selecting a domain in which the encoding is performed for each predetermined band, the bitstream can be efficiently encoded and transmitted and received, and compatibility can be provided.
摘要:
An audio signal processing apparatus and method and a computer readable recording medium storing a computer program for the method are provided. The audio signal processing apparatus includes: an input unit that receives the audio signal; and a signal processing unit that processes the audio signal received from the input unit using at least one of network information and terminal information and signal information, wherein the network information refers to information regarding the network, the status of the network varies at any time, the terminal information refers to information regarding the terminal, the status of the terminal varies at any time, and the signal information refers to information on the audio signal. The audio signal can be efficiently streamed in real-time using the network information and/or the terminal information, which vary at any time, so that the audio signal transmitted from, for example, a server side, can be seamlessly received by a terminal and can be reproduced at optimal, high sound quality by the terminal.
摘要:
A noise filling method is provided that includes detecting a frequency band including a part encoded to 0 from a spectrum obtained by decoding a bitstream; generating a noise component for the detected frequency band; and adjusting energy of the frequency band in which the noise component is generated and filled by using energy of the noise component and energy of the frequency band including the part encoded to 0.
摘要:
A bit allocating method is provided that includes determining the allocated number of bits in decimal point units based on each frequency band so that a Signal-to-Noise Ratio (SNR) of a spectrum existing in a predetermined frequency band is maximized within a range of the allowable number of bits for a given frame; and adjusting the allocated number of bits based on each frequency band.