摘要:
An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.
摘要:
A sound reproduction system for automatically conducting a tone control for an audio signal according to the present invention includes: an input terminal for inputting an audio signal; a judgment circuit for determining a ratio between a stereophonic signal component and a monophonic signal component of the input audio signal and for outputting a judgment signal; a signal processing section for receiving the input audio signal and processing the input audio signal so as to generate an output signal having uniform transmission characteristics irrespective of the listening position; an adder for receiving the input audio signal and the output signal and for adding the input audio signal with the output signal at a certain addition ratio based on the judgment signal so as to generate an added signal; and a loudspeaker for receiving the added signal and for reproducing the added signal in a plurality of positions.
摘要:
A sound field controller for generating apparent sound sources by adjusting the amplitude and delay time of a sound signal so that the sound will be perceived by plural listeners as sound coming from a location separated from the specific location of the front speakers, and for additionally controlling the effect of the apparent sound sources by evaluating the attributes of the source sound signal. The controller includes FIR filters for generating a left sound pattern signal, FIR filters for generating a right sound pattern signal, a first delay circuit for delaying the left and right sound pattern signals by a first predetermined time and applying the delayed left and right sound pattern signals to the left and right speakers, respectively, to introduce an apparent sound source located left rear of a center listener; and a second delay circuit for delaying the left and right sound pattern signals by a second predetermined time and applying the delayed left and right sound pattern signals to the right and left speakers, respectively, to introduce an apparent sound source located right rear of a center listener.
摘要:
To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.A stream synthesizing device includes an input unit which inputs at least two coded signals each including a first downmix acoustic signal and an extended signal, each of first downmix acoustic signals being obtained by coding an acoustic signal into which at least two sound signals are downmixed, and the extended signal being for obtaining the at least two sound signals out of the first downmix acoustic signal; a coded signal generating unit which generates: a second downmix acoustic signal and an extended signal based on each of coded signals inputted by the input unit, the second downmix acoustic signal being for obtaining each of the first downmix acoustic signals, and the generated extended signal being for obtaining each of the first downmix acoustic signals out of the second downmix acoustic signal; and generate a coded signal including the generated second downmix acoustic signal, the generated extended signal, and each of extended signals included in the corresponding inputted coded signal; and an output unit which outputs the generated coded signal.
摘要:
To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.A stream synthesizing device includes an input unit which inputs at least two coded signals each including a first downmix acoustic signal and an extended signal, each of first downmix acoustic signals being obtained by coding an acoustic signal into which at least two sound signals are downmixed, and the extended signal being for obtaining the at least two sound signals out of the first downmix acoustic signal; a coded signal generating unit which generates: a second downmix acoustic signal and an extended signal based on each of coded signals inputted by the input unit, the second downmix acoustic signal being for obtaining each of the first downmix acoustic signals, and the generated extended signal being for obtaining each of the first downmix acoustic signals out of the second downmix acoustic signal; and generate a coded signal including the generated second downmix acoustic signal, the generated extended signal, and each of extended signals included in the corresponding inputted coded signal; and an output unit which outputs the generated coded signal.
摘要:
An audio signal coding apparatus includes a first-stage encoder for quantizing the time-to-frequency transformed audio signal and second-and-subsequent-stages of encoders each for quantizing a quantization error output from the previous-stage encoder A characteristic decision unit is provided which decides the frequency band of an audio signal to be quantized by each encoder of multiple-stage encoders, and a coding band control unit receives the frequency band decided by the characteristic decision unit and the time-to-frequency transformed audio signal, decides the order of connecting the respective encoders, and transforms the quantization bands of the encoders and the connecting order to code sequences. Therefore, it is possible to provide an audio signal coding apparatus performing adaptive scalable coding, which exhibits sufficient performance when coding various audio signals.
摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要:
An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.
摘要:
An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.