Signal processor

Details Signal processor Signal processor

2000-06-30

2004-06-15

Hudspeth, David (Department: 2651)

Dynamic information storage or retrieval

Binary pulse train information signal

Including sampling or a/d converting

C341S061000, C375S240110

Reexamination Certificate

active

06751177

ABSTRACT:

TECHNICAL FIELD
The present invention relates to a signal processing apparatus and, more particularly to a signal processing apparatus which reduces operation amount in decoding when reproducing a plurality of signals of different sampling frequencies.
BACKGROUND ART
According to a DVD audio standard for a DVD-based audio disc, its storage capacity is 4.7 GB. DVD audio recording scheme is PCM, like CD or DVD-ROM and, as for its specification, a sampling frequency indicative of fidelity to original sound with which audio is recorded is 192 KHz at maximum, which is about 4.3 times as high as that (44.1 KHz) of the CD. This enables to record audio of highest quality.
FIG. 3
is a diagram showing conception of a sound field formed by reproduced DVD audio. In the figure, reference numeral
40
denotes a center speaker, and
41
a
and
41
b
denote left and right speakers placed at the left of the center speaker
40
and at the right of the same, respectively. Reference numerals
42
a
and
42
b
denote left and right surround speakers placed behind an auditor, for increasing realism, and
43
denotes a speaker called “sub-woofer”, for outputting relatively low sound. According to the DVD audio standard, reproduction can be performed by using 6 speakers (6 channels) even when sampling frequencies and the numbers of quantization bits of respective channels differ from each other. For instance, in configuration shown in
FIG. 3
, the center speaker
40
and the left and right channel speakers
41
a
and
41
b
for which relatively high sound quality is demanded, perform reproduction at 96 KHz, while the left and right surround speakers
42
a
and
42
b
and the sub-woofer
43
for which relatively high sound quality is not demanded, perform reproduction at 48 KHz.
By the way, when data of respective channels are to be recorded at 96 KHz and in 24 bits for data of 6 channels, a standard for a maximum transfer rate would be exceeded. Accordingly, it becomes necessary to compress data when recorded. A compression method includes irreversible compression using a psychoacoustic model for use in MPEG or AC3, and “Lossless compression” which is capable of completely restoring data to the state before compression by employing entropy coding as reversible compression, such as Huffman coding. In order to reproduce audio of high quality with fidelity to original sound as described above, the Lossless compression is desirably employed. This enables to reproduce audio of high quality of 6 channels at 96 KHz and in 24 bits, in data transfer of the DVD audio. On the other hand, even when the standard for the maximum transfer rate is not exceeded, the Lossless compression enables to record data of 4.7 GB for a long time period.
FIG. 4
is a block diagram showing a conventional DVD audio recording apparatus. For the sake of simplicity, 3 channels are illustrated, although 6 channels are actually used. In
FIG. 4
, reference numerals
51
a
and
51
b
denote upsampling means which receive signals of the channels
2
and
3
at sampling frequencies of 48 KHz, and adapt their respective sampling frequencies to 96 KHz for the channel
1
. Reference numeral
50
denotes a timing delay unit for delaying the signal of the channel
1
while the signals of the channel
2
and
3
are upsampled, and
52
denotes a filter circuit for filtering the upsampled signals of the channels
2
and
3
and performing interpolation for them so that they are smoothed. Reference numeral
54
denotes Lossless compression means for performing reversible compression of the signals of the channels
2
and
3
which passed through the filter circuit
52
and the signal of the channel
1
delayed by the delay unit
50
. Reference numeral
53
denotes format transformation means for transforming a Lossless-compressed signal into data having a predetermined format which can be written to a recording medium
56
, and
55
denotes recording means for recording the compressed data into the recording medium
56
.
To upsample the sampling frequencies of the signals of the channels
2
and
3
from 48 KHz to 96 KHz, respectively, with the above-mentioned construction, the upsampling means
51
a
and
51
b
insert a predetermined number of “zeros” into data so that the sampling frequencies are twice higher (48×2=96), and then a filter circuit
52
having a given factor in a subsequent stage replaces the inserted “zero” samples with :samples used for smooth interpolation. To be specific, the upsampling process for inserting the samples having “zero” values into the data of the signals of the channels
2
and
3
is performed so that the sample having the “zero” value is placed in every other sample of the data. While the signals of the channels
2
and
3
are upsampled, the signal of the channel
1
is delayed by the delay unit
50
. Instead of the above “0” insertion, processing performed by the upsampling means
51
a
and
51
b
may be a sample holding process which holds a predetermined number of previous sample data or an interpolation process using straight lines rather than “zeros”. Here, in the sample holding process, the data of the signals of channels
2
and
3
are interpolated so that after each of the samples constituting that data, a sample having the same value as that sample is placed. For the filter circuit
52
, a low pass filter can be realized by a filter such as an FIR (Filter Impulse Response) or an IIR (Infinite Impulse Response). The filter circuit
52
filters the signals output from the upsampling means
51
a
and
51
b
by using the above filter.
The outputs of the filter circuit
52
and the output of the delay unit
50
are processed by the Lossless compression means
54
and then processed by the format transformation means
53
, and the resulting data is written to the recording medium (DVD audio disc)
56
by using the recording means
55
.
To read so created data from the recording medium
56
to reproduce audio, a reproducing apparatus shown in
FIG. 5
is used. In
FIG. 5
, reference numeral
60
denotes reading means for reading data from the recording medium
56
, and reference numeral
62
denotes format inverse-transformation means for transforming the read data (Lossless-compressed) into a signal (Lossless-compressed) having a format of reproducible audio signal. Reference numeral
61
denotes compressed-data decompression means for decompressing the data (Lossless-compressed) which has been subjected to the format inverse-transformation, and reference numeral
63
denotes a filter circuit for downsampling predetermined decompressed data as required.
To reproduce the predetermined data decompressed by the compressed-data decompression means
61
at a sampling frequency of 48 KHz downsampled from 98 KHz, with the above-described construction, the filter circuit
63
downsamples this data.
In the conventional signal processing apparatus so constructed, the filter circuit temporarily equalizes the sampling frequencies of the plurality of signals at a recording time, to be recorded in the recording medium, while the filter circuit at a reproducing end changes the sampling frequencies of the predetermined channels into the predetermined sampling frequencies, to output the signals.
In this case, when high precision is required for the filter circuits used in the above processing, the amount of operation therein is noticeably increased, and burden on hardware is correspondingly increased. In addition, the processed signals are reproduced unsatisfactorily.
The present invention is directed to solving the above problem, and an object of the present invention is to provide a signal processing apparatus which is capable of reducing operation amount in filter circuits when processing a plurality of signals of different sampling frequencies, and reproducing all the signals completely.
DISCLOSURE OF THE INVENTION
According to aspect
1
of the present invention, there is provided a signal processing apparatus for encoding a plurality of channel signals of different sampling frequencies to be recorded

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