摘要:
A method and apparatus to scalably encode and/or decode an audio signal includes encoding a specific band signal included in an input signal, encoding a frequency envelope of an excited signal in which the encoded specific band signal is removed from the input signal, encoding a residual signal in which the encoded frequency envelope is removed from the excited signal, and forming a bit-stream by scalably packing the encoded specific band signal, frequency envelop, and residual signal.
摘要:
A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data. The method may further include interpolating frequency amplitude data that corresponds to a second one of the even-numbered or odd-numbered frequency amplitude that is not quantized from among the normalized frequency amplitude data using the quantized first one of the even-numbered or odd-numbered data, and quantizing an interpolation error corresponding to a difference between the second frequency amplitude data that is not quantized and the interpolated frequency amplitude data.
摘要:
A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data. The method may further include interpolating frequency amplitude data that corresponds to a second one of the even-numbered or odd-numbered frequency amplitude that is not quantized from among the normalized frequency amplitude data using the quantized first one of the even-numbered or odd-numbered data, and quantizing an interpolation error corresponding to a difference between the second frequency amplitude data that is not quantized and the interpolated frequency amplitude data.
摘要:
Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.
摘要:
Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.
摘要:
A method, apparatus, and medium for classifying a speech signal and a method, apparatus, and medium for encoding the speech signal using the same are provided. The method for classifying a speech signal includes calculating classification parameters from an input signal having block units, calculating a plurality of classification criteria from the classification parameters, and classifying the level of the input signal using the plurality of classification criteria. The classification parameters include at least one of an energy parameter of the input signal, a cross-correlation parameter between a specific block of a present frame and the input signal, and an integrated cross-correlation parameter obtained by accumulating the cross-correlation parameter.
摘要:
Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.
摘要:
Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.
摘要:
A method, apparatus, and medium for classifying a speech signal and a method, apparatus, and medium for encoding the speech signal using the same are provided. The method for classifying a speech signal includes calculating classification parameters from an input signal having block units, calculating a plurality of classification criteria from the classification parameters, and classifying the level of the input signal using the plurality of classification criteria. The classification parameters include at least one of an energy parameter of the input signal, a cross-correlation parameter between a specific block of a present frame and the input signal, and an integrated cross-correlation parameter obtained by accumulating the cross-correlation parameter.
摘要:
Audio coding and decoding apparatuses and methods that can optimize the quality of an audio signal including harmonics, and recording mediums storing the methods. An audio coding apparatus includes: a first harmonic coding module performing first harmonic coding on an input audio signal using a pitch lag of the input audio signal and producing a quantized linear prediction coding coefficient; a first detector detecting a first difference audio signal from a difference between an audio signal output from the first harmonic coding module and the input audio signal; a second harmonic coding module performing harmonic coding on the first difference audio signal using the quantized linear prediction coding coefficient and a previous harmonic coding result; a second detector detecting a second difference audio signal obtained from a difference between an audio signal output from the second harmonic coding module and the first difference audio signal; and a code excited linear prediction (CELP) module CELP coding the second difference audio signal using the quantized linear prediction coding coefficient obtained from the first harmonic coding module.