摘要:
Activity is determined for each frequency band in a frame, and when it is determined that an activity-OFF state has not continued for a predetermined number of times for preceding frames, normal coding processing is performed for the frequency band. When it is determined that the activity-OFF state has continued for the predetermined number of times or more, DTX coding is performed for the frequency band. After this processing has been performed for all of the bands of one frame, a total power of the one entire frame and the power of the band or bands to which the DTX coding is applied are calculated. Subsequently, a new target bit value is calculated based on a ratio of the total power of the one entire frame and the power of the band or bands to which the DTX coding is applied.
摘要:
According to an aspect of the invention, there is provided an audio data processing apparatus including: a decoding unit configured to decode audio encoding data, while the decoding unit switches an M/S stereo application mode and an M/S stereo non-application mode, and thereby outputting frequency domain audio data; an inverse quantizing unit configured to inversely quantize and output the frequency domain audio data; an M/S stereo judgment unit configured to decide whether or not the M/S stereo application mode is applied to the scale factor band, and extract and output a frequency domain audio data of the S channel at a part of scale factor band to which the M/S stereo application mode is applied, and generate and output a frequency domain audio data of the S channel at a part of scale factor band to which the M/S stereo application mode is not applied.
摘要:
Activity is determined for each frequency band in a frame, and when it is determined that an activity-OFF state has not continued for a predetermined number of times for preceding frames, normal coding processing is performed for the frequency band. When it is determined that the activity-OFF state has continued for the predetermined number of times or more, DTX coding is performed for the frequency band. After this processing has been performed for all of the bands of one frame, a total power of the one entire frame and the power of the band or bands to which the DTX coding is applied are calculated. Subsequently, a new target bit value is calculated based on a ratio of the total power of the one entire frame and the power of the band or bands to which the DTX coding is applied.
摘要:
A wideband speech coding apparatus comprises a wideband speech coding unit configured to code an input speech signal at a bit rate which is set in advance in accordance with a wideband speech signal. The apparatus further comprises an identification unit configured to identify whether the input speech signal is a wideband speech signal or a narrowband speech signal. The apparatus further comprises a control unit configured to cause the wideband speech coding unit to code the input speech signal, when the identification unit identifies that the input speech signal is the wideband speech signal, and to cause the wideband speech coding unit to raise the bit rate to code the input speech signal, when the identification unit identifies that the input speech signal is the narrowband speech signal. The apparatus further comprises an output unit to output the input speech signal coded by the wideband speech coding unit.
摘要:
A signal bandwidth extension apparatus includes a determination unit which determines whether or not a peak component of the input signal is lacked in the band to be extended, and a control unit which controls to extend the bandwidth when the determination unit determines that the peak component of the input signal is lacked in the band to be extended, and not to extend the bandwidth when the determination unit determines that the peak component is not lacked.
摘要:
According to one embodiment, an acoustic signal compensator includes an acoustic signal receiving module, a compensator, and an output module. The acoustic signal receiving module receives an acoustic signal. The compensator performs compensation on the acoustic signal, as compensation of acoustic characteristics of an ear including an ear canal having a first-order resonance characteristic and a second-order resonance characteristic, to suppress a first-order frequency of ear resonance and a second-order frequency lower than double of the first-order frequency. The output module outputs the acoustic signal compensated by the compensator.
摘要:
A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.
摘要:
A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.
摘要:
A common component extraction unit extracts a component Mi(n) included commonly in the plural input channels. A non-common component extraction unit extracts a component Si(n) not common to the plural input channels. A common component noise suppression unit obtains signal Mo(n) by executing a noise suppression process for the component Mi(n). A non-common component processing unit obtains signal So(n) by executing attenuation or emphasis for the component Si(n). A plural-channel generation unit removes noise of the common component from signals In(n, k) of the plural input channels, by using the signals Mo(n) and So(n) such that the non-common component is attenuated or emphasized.
摘要:
A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.