摘要:
There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).
摘要:
Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer includes a power correlation calculator which calculates the power of the left channel signal, the power of the right channel signal, and the correlation between the left channel signal and the right channel signal; an intermediate value calculator which calculates the intermediate value which is the difference between left channel signal the power and the right channel signal power; a codebook which holds a plurality of sets of the coefficients related to the transform coefficients of the principal component analysis transform and the code; and a quantizer which calculates the sum of the first multiplication result obtained by multiplying the coefficient by the correlation value and the second multiplication result obtained by multiplying the coefficient by the intermediate value as the cost function E, selects the coefficients where the cost function E becomes the maximum, and fetches the code related to the selected coefficients as the quantized code.
摘要:
Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.
摘要:
There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).
摘要:
Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.
摘要:
Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer (110) is comprised of a power correlation calculation unit (111) which calculates the power (C11) of the left channel signal, the power (C22) of the right channel signal, and the correlation (C12) between the left channel signal and the right channel signal; an intermediate value calculation unit (112) which calculates the intermediate value (C1122) which is the difference between the power (C11) and the power (C22); a codebook (113) which holds a plurality of sets of the coefficients ?1,n,?2,n related to the transform coefficients of the principal component analysis transform and the code; and a quantizer (114) which calculates the sum of the first multiplication result obtained by multiplying the coefficient ?1,n by the correlation value C12 and the second multiplication result obtained by multiplying the coefficient ?1,n by the intermediate value C1122 as the cost function E, selects the coefficients ?1,n,?2,n where the cost function E becomes the maximum, and fetches the code related to the selected coefficients ?1,n,?2,n as the quantized code.
摘要:
A CELP speech decoder includes an adaptive codebook that generates an adaptive code vector and a random codebook that generates a random code vector. The random codebook includes an input vector provider that provides an input vector including at least one pulse, each pulse having a position and a polarity, a fixed waveform storage that stores at least one fixed waveform, and a selector that selects at least one of a first process and a second process based on a value of an adaptive codebook gain. The random codebook further includes a convolution section that generates the random code vector by convoluting the at least one fixed waveform with the input vector when the first process is selected. A synthesis filter outputs synthesized speech by performing linear prediction coefficient synthesis on a signal based on the adaptive code vector and the random code vector.
摘要:
A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.
摘要:
A noise canceller removes a noise component from an input speech signal. The noise canceller includes a noise cancellation coefficient adjuster that adjusts a noise cancellation coefficient to determine an amount of noise cancellation. A noise spectrum storage device stores an estimated noise spectrum. A noise estimator estimates a noise spectrum by comparing an input spectrum with a noise spectrum stored in the noise spectrum storage device. A noise canceling/spectrum compensator subtracts the noise spectrum stored in the noise spectrum storage device from the input spectrum based on a coefficient acquired by the noise cancellation coefficient adjuster.
摘要:
A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.