摘要:
There is disclosed an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal. In this device, a first encoding section (115) subjects an input signal (S11) to audio encoding processing of the CELP method and outputs the obtained first encoded information (S12) to a parameter decoding section (120). The parameter decoding section (120) acquires a first quantization LSP code (L1), a first adaptive excitation lag code (A1), and the like from the first encoded information (S12), obtains a first parameter group (S13) from these codes, and outputs it to a second encoding section (130). The second encoding section (130) subjects the input signal (S11) to a second encoding processing by using the first parameter group (S13) and obtains second encoded information (S14). A multiplexing section (154) multiplexes the first encoded information (S12) with the second encoded information (S14) and outputs them via a transmission path N to a decoding apparatus (150).
摘要:
An excitation vector generator includes an input vector providing system that is capable of providing an input vector having at least one pulse, each pulse having a predetermined position and a respective polarity. A fixed waveform storage system is capable of storing at least one fixed waveform. An arranging system is capable of arranging the at least one fixed waveform in accordance with the position and the polarity of the at least one pulse.
摘要:
First codebook 61 and second codebook 62 respectively have two subcodebooks, and in respective codebooks, addition sections 66 and 67 obtain respective excitation vectors by adding sub-excitation vectors fetched from respective two subcodebooks. Addition section 68 obtains an excitation sample by adding those excitation vectors. According to the aforementioned constitution, it is possible to store sub-excitation vectors with different characteristics in respective sub-codebooks. Therefore, it is possible to correspond to input signals with various characteristics, and achieve excellent sound qualities at the time of decoding.
摘要:
Base layer coding section 101 encodes an input signal to obtain base layer coded information. Base layer decoding section 102 decodes the base layer coded information to obtain a base layer decoded signal and long term prediction information (pitch lag). Adding section 103 inverts the polarity of the base layer decoded signal to add to the input signal, and obtains a residual signal. Enhancement layer coding section 104 encodes a long term prediction coefficient calculated using the long term prediction information and the residual signal to obtain enhancement layer coded information. Base layer decoding section 152 decodes the base layer coded information to obtain the base layer decoded signal and long term prediction information. Using the long term prediction information, enhancement layer decoding section 153 decodes the enhancement layer coded information to obtain an enhancement layer decoded signal. Adding section 154 adds the base layer decoded signal and enhancement layer decoded signal to obtain a speech/sound signal. It is thereby possible to implement scalable coding with small amounts of calculation and coded information.
摘要:
A code excited linear prediction speech decoder is provided. An adaptive codebook generates an adaptive code vector. A random codebook generates a random code vector. A synthesis filter receives a signal based on the adaptive code vector and the random code vector, and performs linear prediction coefficient synthesis on the signal. The random codebook includes a pulse vector provider that provides a pulse vector having a signed unit pulse, a comparator that compares a value of adaptive codebook gain with a preset threshold value, a selector that selects a dispersion pattern from a plurality of dispersion patterns stored in a memory in accordance with a result of the comparison, and a generator that generates the dispersed vector by convoluting the pulse vector and the selected dispersion pattern.
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.
摘要:
A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained. Accordingly, because an appropriate coding module can be easily selected from a plurality of coding modules according to the coding module selecting rule, any allophone occurring in a reproduced speech can be prevented at a low calculation volume.
摘要:
A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained. Accordingly, because an appropriate coding module can be easily selected from a plurality of coding modules according to the coding module selecting rule, any allophone occurring in a reproduced speech can be prevented at a low calculation volume.