摘要:
Provided are a vector quantization device, a voice coding device, a vector quantization method, and a voice coding method which enable a reduction in the calculation amount of voice codec without deterioration of voice quality. In the vector quantization device, a first reference vector calculation unit (201) calculates a first reference vector by multiplying a target vector (x) by an auditory weighting LPC synthesis filter (H), and a second reference vector calculation unit (202) calculates a second reference vector by multiplying an element of the first reference vector by a filter having a high pass characteristic. A polarity preliminary selection unit (205) generates a polar vector by disposing a unit pulse having a positive or negative polarity, which is selected on the basis of the polarity of an element of the second reference vector, in the position of said element.
摘要:
Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (111) having: a section search unit (121) which searches for a pulse for each of bands into which a predetermined search section is divided; and a whole search unit (122) which performs search for a pulse over the entire search section. The shape of an input spectrum is quantized by a small number of pulse positions and polarities. A gain quantization unit (112) calculates a gain of the pulse searched by the shape quantization unit (111) and quantizes the gain for each of the bands.
摘要:
Provided are a coding device, a communication processing device, and a coding method, whereby processing operation load (computational load) is significantly reduced for a configuration which computes either frame energy or sub-frame energy of an input signal, using auto-correlation operations, without causing a decline in the precision of either the frame energy or the sub-frame energy. In a coding device (101), a sub-frame energy computation unit (201) computes the sub-frame energy by substituting the sum of input signal auto-correlation operations in a first range with the sum of auto-correlation operations in a second range which differs at least partially from the first range.
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:
An encoding device and an encoding method are provided for encoding by reducing the number of samples to be processed when encoding higher-band spectrum data according to lower-band spectrum data in a wide-band signal. The device and the method can obtain a high-quality decoded signal even if a large quantization distortion is caused in the lower-band spectrum data. When encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for a part (a head portion) of the higher-band spectrum data, the lower-band spectrum data after being quantized is subjected to approximate partial search and higher-band spectrum data is generated according to the search result.
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:
An encoder generating a decoded signal with an improved quality by scalable encoding by canceling the characteristic inherent to the encoder and causing degradation of quality of the decoded signal. In the encoder, a first encoding section (102) encodes the input signal after down sampling, a first decoding section (103) decodes first encoded information outputted from the first encoding section (102), an adjusting section (105) adjusts the first decoded signal after up sampling by convoluting the first decoded signal after up sampling and an impulse response for adjustment, an adder (107) inverses the polarity of adjusted first decoded signal and adds the first decoded signal having the inverted polarity to the input signal, a second encoding section (108) encodes the residual signal outputted from the adder (107), and a multiplexing section (109) multiplexes the first encoded information outputted from the first encoding section (102) and the second encoded information outputted from the second encoding section (108).
摘要:
Disclosed are a vector quantization device and others capable of adaptively adjusting a vector space of a code vector for quantization of a second stage by using a quantization result of a first stage and improving the quantization accuracy. In the device, the first quantization unit (101) performs quantization of an LSP vector; a quantization residual difference generation unit (102) acquires a residual difference between the LSP vector and the first quantization vector obtained by the first quantization unit (101) as a quantization residual difference vector; an addition factor selection unit (103) selects one of the addition factor code vectors in an addition factor codebook as an addition factor vector according to the first code obtained by the first quantization unit (101); an addition residual difference generation unit (104) acquires a residual difference between the quantization residual difference vector and the addition factor vector as an addition residual vector; and a second quantization unit (105) performs quantization of the addition residual difference vector.
摘要:
A stereo signal converter capable of realizing encoding with less redundancy, low bit-rate, and high quality even if the positions of sound sources are different from one another. In this device, a sample difference analyzing section (111) uses the signal in which a right-channel signal is shifted by a sample difference (d) in terms of time and a left-channel signal to compute a sample difference (D) in which the correlation becomes highest. A sample difference value computing section (112) computes a sample difference value (z) (the value to shift the right-channel signal in the current frame) on the basis of the value after the right-channel signal is shifted in the previous frame and the sample difference (D). A sample difference value encoding section (113) encodes the sample difference value (z). A slide section (114) shifts the right-channel signal by the sample difference value (z) in terms of time. A sum difference computing section (115) adds the left-channel signal and the shifted right-channel signal to generate a monaural signal and subtracts the shifted right-channel signal from the left-channel signal to generate a side signal.
摘要:
There is provided a vector conversion device for converting a reference vector used for quantization of an input vector so as to improve a signal quality including audio. In this vector conversion device, a vector quantization unit (902) acquires a number corresponding to a decoded LPC parameter of a narrow band from all the code vectors stored in a code book (903). A vector dequantization unit (904) references the number of the code vector obtained by the vector quantization unit (902) and selects a code vector from the code book (905). A conversion unit (906) performs calculation by using a sampling-adjusted decoded LPC parameter obtained from an up-sampling unit (901) and a code vector obtained from the vector dequantization unit (904), thereby obtaining a decoded LPC parameter of a broad band.