Digital audio decoder

Details Digital audio decoder Digital audio decoder

2001-05-23

2004-04-20

Isen, Forester W. (Department: 2644)

Data processing: generic control systems or specific application

Specific application, apparatus or process

Digital audio data processing system

C381S022000, C381S106000, C704S500000, C704S503000

Reexamination Certificate

active

06725110

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to digital audio decoders that decode digital audio signals (or bit stream data) which are compressed by sub-band coding methods such as MPEG/Audio signals, ATRAC signals and AC-3 signals (where ‘MPEG’ stands for ‘Moving Picture Experts Group’, and ‘ATRAC’ stands for ‘Adaptive Transform Acoustic Coding’).
2. Description of the Related Art
Conventionally, there are provided various types of compression methods for compressing digital audio signals, one of which is known as the MPEG/Audio standard.
FIG. 5
shows an example of a data compression circuit based on the aforementioned standard. Input digital audio signals Da are partitioned into blocks (namely, frames), each of which contains a prescribed number of samples. In the data compression circuit shown in
FIG. 4
, the input digital audio signals Da are processed by two paths. A first path brings the digital audio signals Da to a filter bank
1
in which they are divided into sub-band signals of thirty-two bands that have equal bandwidths respectively. Each of the sub-band signals is down-sampled to {fraction (1/32)} of the sampling frequency. Then, the sub-band signals are forwarded to a scale factor extraction normalization circuit
2
, wherein a sample having a maximal absolute value is detected from each frame of the sub-band signals. The detected value is subjected to quantization to produce a specific value, which is called a scale factor. Using the scale factors, the sub-band signals are subjected to division process and are then subjected to normalization into a prescribed range of values within ±1.
A second path brings the digital audio signals Da to an auditory psychology analysis (or auditory perception analysis) block
3
in which frequency spectra are calculated by the fast Fourier transform (FFT). Based on the calculated frequency spectra, the auditory psychology analysis block
3
produces masking thresholds for the sub-band signals respectively, namely allowable quantization noise power. A bit allocation block
4
operates under the restriction of the output of the auditory psychology analysis block
3
and a prescribed number of bits that can be used in one frame, which is determined by the bit rate. Under the aforementioned restriction, the bit allocation block
4
performs repeated loop processes to determine numbers of quantized bits (hereinafter, referred to as ‘quantization bit numbers’) with respect to sub-bands respectively. Using the quantization bit numbers set for the sub-bands respectively, the quantization block
5
performs quantization on the sub-band signals output from the scale factor extraction normalization circuit
2
. That is, the quantization block
5
produces ‘quantized’ sub-band samples. A bit stream generation block
6
combines the quantized sub-band samples, bit allocation information and scale factor for each of the sub-bands together in a multiplexing manner. In addition, a header is added to them to create a bit stream, which is output from the bit stream generation block
6
.
FIG. 6
shows an example of a configuration of a decoder (or data expansion circuit) that decodes the bit stream, which is produced by the data compression circuit of FIG.
4
. Herein, a bit allocation information and scale factor extraction block
11
extracts the bit allocation information and scale factor from the bit stream. In response to the bit allocation information, an inverse quantization circuit
12
reads bit strings respectively corresponding to thirty-two sub-band samples from the bit stream, wherein the bit strings are subjected to inverse quantization with respect to each of the sub-band samples and are then subjected to multiplication by the scale factors. Thus, the inverse quantization circuit
12
produces ‘inversely quantized’ sub-band signals, which are synthesized together to reproduce the original digital audio signals by a sub-band synthesis filter bank
13
.
Recently, so-called digital sound sources based on the MPEG/Audio standard are widely used in a variety of fields such as pinball game machines, which are widely used in amusement places in Japan.
FIG. 6
shows a configuration of a musical tone generation circuit that operates based on the MPEG/Audio standard. Herein, reference numerals
21
designate MPEG/Audio sound sources that contain memories for storing musical tone data, which are made in forms of bit streams respectively, and readout circuits for reading data from the memories respectively. Reference numerals
22
designate decoders (see
FIG. 5
) that expand output data of the MPEG/Audio sound sources to restore original PCM musical tone data (where ‘PCM’ stands for ‘PulseCode Modulation’). Reference numerals
23
designate multipliers that perform gain controls on outputs of the decoders
22
. Reference numeral
24
designates an adder that adds together outputs of the multipliers
23
. The above describes an example of the configuration of the musical tone generation circuit that is applied to the pinball game machine, for example. This musical tone generation circuit normally provides plural sound sources for multiple channels. That is, the plural sound sources produce MPEG/Audio digital musical tone signals, which are synthesized together to form composite musical tone signals.
The decoder
22
shown in
FIG. 6
has processes regarding inverse quantization and sub-band synthesis filter bank, wherein the sub-band synthesis filter bank
13
is configured by a RAM having a relatively large storage capacity. For this reason, the aforementioned musical tone generation circuit of the MPEG/Audio standard, which provides the decoders
22
subsequently to the sound sources
21
, bears a problem because the total storage capacity should be increased so much.
It is well known that the conventional digital audio devices use so-called bass boost circuits that amplify low-frequency components of sound. The musical tone generation circuit of the MPEG/Audio standard additionally provides bass boost circuits subsequently to the decoders
22
. However, such a configuration causes a problem due to complexity of circuitry because the bass boost circuits should be provided independently of the decoders
22
.
In the fields of the digital audio techniques in these days, so-called surround effect techniques are frequently used to enhance richness of sounds.
FIG. 8
shows an example of a sound effect circuit, which inputs left-channel signals Li and right-channel signals Ri. Herein, a subtracter
25
produces difference signals between the left-channel signals Li and right-channel signals Ri. A low-pass filter (LPF) filters low frequency components of the difference signals, which are applied to multipliers
26
,
27
respectively. The multiplier
26
multiplies them by a positive multiplication coefficient ‘a’, while the multiplier
27
multiplies them by a negative multiplication coefficient ‘−a’. An adder
28
adds together the output of the multiplier
26
and the left-channel signals Li, while an adder
29
adds together the output of the multiplier
27
and the right-channel signals Ri. Thus, the surround effect circuit outputs surround-effect imparted left-channel signals Lo and surround-effect imparted right-channel signals Ro.
It is possible to realize surround effects on musical tone signals of multiple channels. In that case, the musical tone signals are mixed together over the multiple channels with respect to the left channel and right channel respectively. This provides uniform surround effects on all of the channels. However, this is disadvantageous in the prescribe case where one channel is given monaural signals while another channel (left or right channel) is given stereophonic signals because the aforementioned surround effect circuit mistakenly produces mixed signals of two channels as Lo and Ro in FIG.
8
.
Conventionally, a variety of configurations and techniques are proposed for processing of digital audio data. For example, Japanese Patent Unexamined Publication No. Hei 8

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