Content supply system and information processing method

Coded data generation or conversion – Digital code to digital code converters

Reexamination Certificate

Rate now

  [ 0.00 ] – not rated yet Voters 0   Comments 0

Details

C714S758000, C341S173000

Reexamination Certificate

active

06794996

ABSTRACT:

TECHNICAL FIELD
This invention relates to a content supply system and an information processing method for supplying contents such as audio signals or image signals, and particularly to a content supply system and an information processing method in which signals are coded to enable trial viewing/listening and therefore reproduction and recording with high quality is made possible by adding a small quantity of data when a trial viewer/listener decides to purchase the signals.
BACKGROUND ART
A content (software) distribution method is known in which, for example, an acoustic signal or the like is encrypted and then broadcast or recorded to a recording medium so that only a person who purchased a key for decryption is permitted to listen to the signal.
As an encryption method, for example, a method is known in which an initial value of a random-number sequence is given as a key signal for a bit string of a PCM acoustic signal and then a bit string obtained by taking an exclusive OR between the generated random-number sequence of 0/1 and the PCM bit string is transmitted or recorded to a recording medium. As this method is used, a person who acquired the key signal can correctly reproduce the acoustic signal and a person who did not acquire the key signal can only reproduce noise. Of course, it is also possible to use a more complicated method such as so-called DES (Data Encryption Standard) as an encryption method. Description of the DES standard is disclosed in “Federal Information Processing Standards Publication 46, Specifications for the DATA ENCRYPTION STANDARD, 1977, January 15.”
On the other hand, a method for compressing an acoustic signal and then broadcasting or recording the compressed acoustic signal to a recording medium is popularized, and recording media which enable recording of a coded audio signal or the like, such as a magneto-optical disc, are broadly used.
There are various techniques for high-efficiency coding of an audio signal, voice signal, or the like. For example, such techniques may include subband coding (SBC), which is a non-blocked frequency band division system for dividing an audio signal or the like on the time axis into a plurality of frequency bands without blocking and then coding the band-divided audio signal, and so-called transform coding, which is a blocked frequency band division system for transforming (spectrally transforming) a signal on the time axis to a signal of the frequency axis, then dividing the signal into a plurality of frequency bands and coding the signal of each band. Moreover, a high-efficiency coding technique combining the above-described subband coding with transform coding is considered. In that case, for example, after frequency band division is carried out by the above-described subband coding, the signal of each band is spectrally transformed to a signal on the frequency axis and the spectrally transformed signal of each band is coded.
As a filter for the above-described technique, for example, a QMF filter is used. The QMF filter is described in “R. E. Crochiere, Digital coding of speech in subbands, Bell Syst. Tech. J. Vol.55, No.8, 1976.” Moreover, a filter division technique with equal bandwidth is disclosed in “Joseph H. Rothweiler, Polyphase Quadrature Filters—A new subband coding technique, ICASSP 83, BOSTON.”
As the above-described spectral transform, for example, the time axis is transformed to the frequency axis by blocking an input audio signal by predetermined unit time (frame) and then performing discrete Fourier transform (DFT), discrete cosine transform (DCT), modified discrete cosine transform (MDCT) or the like on each of the blocks. MDCT is described in “J. P. Princen, A. B. Bradley, Univ. of Surrey Royal Melbourne Inst. of Tech., Subband/Transform Coding Using Filter Band Designs Based on Time Domain Aliasing Cancellation, ICASSP, 1987”.
If the above-described DFT or DCT is used as a method for transforming a waveform signal to the spectrum, M independent real-number data are provided by performing transform on a time block consisting of M samples. To reduce the connection distortion between time blocks, each time block is usually overlapped with both adjacent blocks by M1 samples each. Therefore, on average, M real-number data are quantized and coded for (M-M1) samples in DFT or DCT.
On the other hand, if the above-described MDCT is used as a method for transforming a waveform signal to the spectrum, M independent real-number data are provided from 2M samples as a result of overlapping both adjacent time blocks by M samples each. Therefore, on average, M real-number data are quantized and coded for M samples in MDCT. A decoding device can reconstruct the waveform signal by performing inverse transform on each block of the code obtained by using MDCT and then adding the resulting waveform elements while letting them interfere with each other.
Generally, by elongating a time block for transform, the frequency resolution of the spectrum is enhanced and the energy concentrates at a specific spectral component. Therefore, by using MDCT in which each block is overlapped with both adjacent blocks by half each to perform transform with a longer block length and in which the number of resulting spectral signals is not increased from the number of the original time samples, more efficient coding can be carried out than when DFT or DCT is used. Moreover, by having each block have a sufficient long overlap with the adjacent blocks, the distortion between the blocks of the waveform signal can be reduced.
By quantizing the signal thus divided to each band by the filter or spectral transform, a band where quantization noise is generated can be controlled and more auditorily efficient coding can be performed by utilizing characteristics such as masking effect. By carrying out normalization for each band with the maximum value of absolute values of signal components in the band before performing quantization, more efficient coding can be performed.
The frequency division width in the case of quantizing each frequency component obtained by frequency band division is determined, for example, in consideration of the human auditory characteristic. Specifically, an audio signal may be divided into a plurality of bands (for example 25 bands) with broader bandwidths for higher-frequency bands which are generally called critical bands. When coding data of each band in this case, coding is carried out by using predetermined bit distribution to each band or adaptive bit allocation to each band. For example, when coding coefficient/factor data resulting from the above-described MDCT processing by the above-described bit allocation, the MDCT coefficient/factor data of each band resulting from MDCT of each of the blocks is coded by using an adaptive number of allocated bits.
For such bit allocation, the following two techniques are known. Specifically, “R. Zelinski and P. Noll, Adaptive Transform Coding of Speech Signals, IEEE Transactions of Acoustics, Speech, and Signal Processing, vol.ASSP-25, No.2, August 1977”, discloses bit allocation based on the magnitude of a signal of each band. In this system, though the quantization noise spectrum is flat and the noise energy is minimum, the actual perception of noise is not optimum because the auditory masking effect is not utilized. “M. A. Kransner, MIT, The critical band coder—digital encoding of the perceptual requirements of the auditory system, ICASSP 1980”, discloses a technique in which a necessary signal-to-noise ratio for each band is obtained using auditory masking, thus performing fixed bit allocation. With this technique, however, even when measuring characteristics by using a sine wave input, a satisfactory characteristic value is not obtained because of fixed bit allocation.
To solve these problems, a high-efficiency coding device is proposed in which all the bits that can be used for bit allocation are divisionally used for a fixed bit allocation pattern predetermined for each small block and for bit allocation dependent on the magnitude of the signal of each bl

LandOfFree

Say what you really think

Search LandOfFree.com for the USA inventors and patents. Rate them and share your experience with other people.

Rating

Content supply system and information processing method does not yet have a rating. At this time, there are no reviews or comments for this patent.

If you have personal experience with Content supply system and information processing method, we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Content supply system and information processing method will most certainly appreciate the feedback.

Rate now

     

Profile ID: LFUS-PAI-O-3240576

  Search
All data on this website is collected from public sources. Our data reflects the most accurate information available at the time of publication.