The present invention relates to sound quality adjustment devices for various audio apparatus and television receivers.
Among examples of the sound quality adjustment devices for various audio apparatus are those disclosed in Japanese Patent Publication Nos. 3206271 and 3329050.
However, with the conventional sound quality adjustment device of
In view of the foregoing, it is an object of the present invention to provide an improved sound quality adjustment device capable of high-speed response to a sound signal level variation.
In order to accomplish the above-mentioned object, the present invention provides a sound quality adjustment device which, for at least one of sound signals of multiple channels, comprises: a filter circuit that extracts a sound signal of a predetermined frequency band from an input sound signal; a boost circuit that performs dynamic range expansion/contraction on the sound signal, extracted by the filter circuit, in accordance with an input level of the sound signal; and an adder that adds together the input sound signal and the sound signal outputted by the boost circuit.
By employing the feed-forward arrangement that performs the dynamic range expansion/contraction on the sound signal in accordance with the level of the sound signal extracted by the filter circuit, the present invention can rapidly respond to a variation in the sound signal level, as compared to the conventionally-known sound quality adjustment device. As a result, even when there has been a rapid increase in the input sound signal level, the present invention can effectively prevent production of an unwanted clipping sound.
Preferably, the sound quality adjustment device further comprises, for the at least one sound signal, a subtracter that subtracts the sound signal, extracted by the filter circuit, from the input sound signal. The adder adds together the subtracted input sound signal and the sound signal outputted by the boost circuit. By the provision of the subtracter that subtracts the filter-extracted sound signal from the input sound signal, the present invention can prevent a dip in the frequency band for which the sound quality adjustment is to be performed and thereby achieve smooth connection among frequency characteristics of the sound.
Preferably, the sound quality adjustment device further comprises, for the at least one sound signal, a decay processing circuit provided, between the filter circuit and the boost circuit, for gradually decaying or attenuating the output level in accordance with lowering of the level of the sound signal extracted by the filter circuit. By the provision of the decay processing circuit between the filter circuit and the boost circuit, the present invention can restrain a too-rapid variation in the level of the sound signal output from the adjustment device, to thereby give a natural auditory sensation to an audience.
Preferably, the sound quality adjustment device further comprises a normalization processing circuit that performs dynamic range expansion/contraction on the sound signal of each of the channels using a same or common gain coefficient corresponding to the greatest level of the sound signals of the multiple channels. With such an arrangement, the present invention can effectively avoid a sound from becoming hard to hear when the sound volume is small while eliminating the inconvenience that the sound becomes too loud when the sound volume is great. Further, the present invention can reduce a difference in sound volume due to differences between audio sources or the like and thereby eliminate the need for frequent sound volume manipulation by the user.
According to another aspect of the present invention, there is provided a sound quality adjustment device, which, for at least one of sound signals of multiple channels, comprises: an extraction section that extracts a sound signal of a predetermined frequency band from an input sound signal; a coefficient calculation section that calculates a filter coefficient on the basis of a level of the sound signal extracted by the extraction section; and a filter processing section that, in accordance with the filter coefficient calculated by the coefficient calculation section, performs a filter process for increasing/decreasing the level of the sound signal of the predetermined frequency band of the input sound signal.
By employing the feed-forward arrangement for changing in real time the filter coefficient of the filter processing section in accordance with the level of the sound signal extracted by the extraction section, the present invention can rapidly respond to a variation in the sound signal level, as compared to the conventionally-known sound quality adjustment device. As a result, even when there has been a rapid increase in the input sound signal level, the present invention can effectively prevent production of an unwanted clipping sound.
Preferably, the sound quality adjustment further comprises, for the at least one sound signal, a decay processing section provided between the extraction section and the filter processing section, the decay processing section gradually attenuating an output level in accordance with lowering of the level of the sound signal extracted by the extraction section. By the provision of the decay processing circuit between the filter circuit and the boost circuit, the present invention can restrain a too-rapid variation in the level of the sound signal output from the adjustment device, to thereby give a natural auditory sensation to the audience.
The following will describe embodiments of the present invention, but it should be appreciated that the present invention is not limited to the described embodiments and various modifications of the invention are possible without departing from the basic principles. The scope of the present invention is therefore to be determined solely by the appended claims.
For better understanding of the objects and other features of the present invention, its preferred embodiments will be described hereinbelow in greater detail with reference to the accompanying drawings, in which:
Behavior of the sound quality adjustment device according to the first embodiment will be described. The LPF 1, which is in the form of an IIR (Infinite Impulse Response) filter, extracts, from an input sound signal (first sound signal), a second sound signal of a bass range lower in frequency than, for example, several hundred Hz. The bass boost circuit 3 performs dynamic range expansion/contraction on the second sound signal, extracted by the LPF 1, in accordance with the input level of the second sound signal.
The HPF 2, which is also in the form of an IIR filter, extracts, from the input first sound signal, a third sound signal of a treble range higher in frequency than, for example, several kHz. The treble boost circuit 4 performs dynamic range expansion/contraction on the third sound signal, extracted by the HPF 2, in accordance with the input level of the third sound signal. Construction of the treble boost circuit 4 is similar to that of the bass boost circuit 3.
The multipliers 5, 6 and 7 multiply the first sound signal, second sound signal output from the bass boost circuit 3 and third sound signal output from the treble boost circuit 4 by respective gain coefficients, to thereby adjust the first to third sound signals to desired levels. The adder 8 adds together the first, second and third sound signals output from the multipliers 5, 6 and 7.
As illustrated in
Further, as shown in
Because the bass components, extracted from the first sound signal, are boosted by the bass boost circuit 3 and then added to the first sound signal as set forth above, the instant embodiment allows the bass-range sound volume in the overall sound volume to approach a predetermined volume level and can thereby impart the sound with “punch” even when the sound source has a small quantity of bass components. Similarly, because the treble components, extracted from the first sound signal, are boosted by the treble boost circuit 4 and then added to the first sound signal as set forth above, the instant embodiment can impart the sound with “modulation” even when the sound source has a small quantity of treble components.
Further, because the instant embodiment employs feed-forward arrangements for performing the dynamic range expansion/contraction of the bass and treble components in accordance with the levels of the bass and treble components extracted by the LPF 1 and HPF 2 instead of employing feedback of the levels of the sound signals having been subjected to the sound quality adjustment, it can rapidly respond to a sound signal level variation, as compared to the conventionally-known sound quality adjustment device shown in
Next, a second embodiment of the present invention will be described.
Thus, the second embodiment can provide the following advantageous benefits in addition to the benefits provided by the first embodiment. With the above-described first embodiment, where the bass and treble components extracted from the first sound signal and boosted by the boost circuits 3 and 4 are added to the bass and treble components originally present in the first sound signal, unnatural dips (sound weakening) may undesirably occur in the bass and treble ranges, which would result in unsmooth connections between frequency characteristics of the sound. To avoid such an inconvenience of the first embodiment, the second embodiment is arranged in such a manner that a bass range is cut out, by the subtracter 9, from the first sound signal while a treble range is cut out, by the subtracter 10, from the first sound signal, so that the first sound signal having passed through the subtracter 10 will have only components of a midrange. Thus, it is possible to prevent bass components of the first and second sound signals from being added together and also prevent treble components of the first and third sound signals from being added together at the time of the addition by the adder 8. In this way, the second embodiment can effectively prevent dips in the bass and treble ranges and thereby achieve smooth connection among frequency characteristics of the sound.
Next, a third embodiment of the present invention will be described.
The decay processing circuit 11 is a circuit for gradually decaying or attenuating the output level in accordance with level lowering of the second sound signal of a bass range extracted by the LPF 1.
When an impulse sound signal IN has been input, the decay processing circuit 11 gradually attenuates the sound signal over a predetermined release time without causing the level of the output signal OUT to follow the sound signal IN that rapidly decreases in level after assuming a maximum value, as illustrated in
In order to realize such a decay process, the sound signal input to the decay processing circuit 11 is sampled at predetermined time intervals, and a comparison is made between a sample value at the current time and an output value at the last sampling time so that the higher of the compared two sample values is selected as an output value at the current time. Thus, when the input sound signal level increases, the latest sample value is constantly selected, so that the output level of the decay processing circuit 11 increases in accordance with the input level. However, when the input sound signal level decreases, the output value at the last sampling time is selected, in which case the value at the last sampling time is attenuated to be set as the output value at the current time. In this case, the decay rate increases with the passage of time as noted above, and the output value is used, in the level comparison at the next sampling, as the output value at the last sampling time.
As stated above, the third embodiment provided with the decay processing circuit 11 can restrain an excessive level variation of the bass range to thereby give a natural auditory sensation to the audience. Although the third embodiment has been described as including the decay processing circuit 11 provided between the LPF 1 and the bass boost circuit 3, such a decay processing circuit may also be provided between the HPF 2 and the treble boost circuit 4. Further, such a decay processing circuit may be applied to the first embodiment as well.
Further, whereas the first to third embodiments have been described only in relation to a sound signal of one channel, the present invention may be applied to sound signals of multiple channels. In such a case, the sound quality adjustment device shown in
The sound quality adjustment may be performed separately for each of the channels, and same or common gain coefficients may be used for these channels. In the case where common gain coefficients are used for all of the channels, the treble boost circuit 4 of each of the sound quality adjustment devices, provided in corresponding relation to the channels Lch, Rch and Cch, may detect, via the level detection section, the greatest level among the treble components of three sound signals of the channels Lch, Rch and Cch extracted by the corresponding HPFs 2, so that gain coefficients corresponding to the greatest level are read out from the gain table. Thus, in the case where the sound quality adjustment is to be performed for the three channels Lch, Rch and Cch, the same level detection circuit and gain table can be shared among the respective treble boost circuits 4 of the three channels although the sound quality adjustment device has to be provided for each of the three channels Lch, Rch and Cch, with the result that the overall circuitry size can be reduced significantly.
Next, a fourth embodiment of the present invention will be described.
The normalization processing circuit 13 includes amplifiers 14-L, 14-R, 14-C, 14-SL, 14-SR and 14-LFE, level detection section 15, and gain table 16. The level detection section 15 detects the greatest level from among sound signals of the channels Lch; Rch and Cch having been subjected to the sound adjustment by the corresponding sound quality adjustment devices 12-L, 12-R and 12-C and sound signals of the other channels SLch, SRch and LFEch that do not pass through the sound quality adjustment devices.
The gain table 16 has prestored therein input sound signal levels and gain coefficients of the amplifiers 14-L, 14-R, 14-C, 14-SL, 14-SR and 14-LFE in association with each other. Gain coefficients corresponding to the greatest level detected by the level detection section 15 are read out from the gain table 16 and supplied to the amplifiers 14-L, 14-R, 14-C, 14-SL, 14-SR and 14-LFE. Each of the amplifiers 14-L, 14-R, 14-C, 14-SL, 14-SR or 14-LFE multiplies the sound signal of the corresponding channel Lch, Rch, Cch, SLch, SRch or LFEch by a gain coefficient output from the gain table 16, to thereby output the multiplied sound signal of the channel Lch, Rch, Cch, SLch, SRch or LFEch.
As in the case of the bass boost circuit 3 or treble boost circuit 4, a plurality of different kinds of input/output characteristics MIN, MID and MAX of the normalization processing circuit 13 may be prepared in advance, and the user may select any one of the MIN, MID and MAX characteristics. In this case, the gain table 16 includes a plurality of tables corresponding to the plurality of different kinds of input/output characteristics.
The fourth embodiment can provide the following advantageous benefits in addition to those provided by the first embodiment. As described above, the fourth embodiment is characterized in that the dynamic range expansion/contraction is performed on the sound signals of the individual channels using common gain coefficients corresponding to the greatest level among the multi-channel sound signals. Thus, when the sound is of a small-volume, the fourth embodiment can turn up the sound volume to prevent the sound from being hidden behind noise, while, when the sound is of a great volume, the fourth embodiment can turn down the sound volume to prevent the sound from becoming offensive to the ears of the audience.
When audio of a motion picture, music or the like is to be reproduced at night, it is common to turn down the sound reproducing volume, so as not to disturb the neighbors. However, if the reproducing volume is turned down, a reproduced sound tends to be hard to hear when a sound signal supplied from an audio apparatus is of a small volume. If the reproducing volume is turned up, on the other hand, a reproduced sound tends to be too loud when a sound signal supplied from an audio apparatus is of a great volume. However, the fourth embodiment arranged in the above-described manner can not only prevent a sound from becoming hard to hear when the reproducing volume is set at a low level, but also prevent a sound from becoming too loud when the reproducing volume is set at a high level. Further, in some case, the use has to frequently manipulate the volume due to, for example, a difference in sound volume between a TV program and commercial or between sound sources. The fourth embodiment can reduce undesired differences in sound volume due to differences between audio sources etc. and thereby eliminate the need for frequent volume manipulation by the user.
The fourth embodiment has been described above as using common gain coefficients for each of predetermined channels. In an alternative, the level detection section is provided separately for each of the channels, and gain coefficients specific to each of the channels may be supplied to the amplifier 14-L, 14-R, 14-C, 14-SL, 14-SR or 14-LFE. In another alternative, the channels are divided into a plurality of groups and the level detection section is provided for each of the groups, and common gain coefficients may be used for each of the groups. For example, the channels may be divided into a group of the channels Lch and Rch and group of the other channels, or into a group of the channels Lch and Rch, group of only the channel Cch and group of the other channels.
Behavior of the quality adjustment device according to the fifth embodiment will be described. The LPF 110, which is in the form of an IIR (Infinite Impulse Response) filter, extracts, from an input sound signal, a sound signal of a bass range lower in frequency than, for example, several hundred Hz. The HPF 120, which is also in the form of an IIR filter, extracts, from the input sound signal, a sound signal of a treble range higher in frequency than, for example, several kHz.
The decay processing section 130 gradually attenuates the output level in accordance with level lowering of the sound signal of the bass range extracted by the LPF 110, while the decay processing section 140 gradually attenuates the output level in accordance with level lowering of the sound signal of the treble range extracted by the HPF 120.
When an impulse sound signal IN has been input, the decay processing section 130, as illustrated in
In order to realize such a decay process, the sound signal input to the decay processing section 130 is sampled at predetermined time intervals, and a comparison is made between a sample value at the current time and an output value at the last sampling time so that the higher of the compared two sample values is selected as an output value at the current time. Thus, as the input sound signal level increases, the latest sample value is constantly selected, but, as the input sound signal level decreases, the output value at the last sampling time is selected, in which case the value at the last sampling time is attenuated to be set as the output value at the current time. In this case, the decay rate increases with the passage of time as noted above, and the output value is used, in the level comparison at the next sampling, as the output value of the last sampling time. The other decay processing section 140 is arranged in a similar manner to the decay processing section 130.
The coefficient calculation section 150 calculates a filter coefficient of the bass boost circuit 170 on the basis of the level of the sound signal output from the decay processing section 130. The coefficient calculation section 160, on the other hand, calculates a filter coefficient of the treble boost circuit 180 on the basis of the level of the sound signal output from the decay processing section 140. For the filter coefficient calculation purposes, tables having prestored therein sound signal levels and filter coefficients in association with each other, for example, may be provided in the coefficient calculation sections 150 and 160 so that particular filter coefficients, corresponding to the levels of the sound signals output from the decay processing sections 130 and 140, are read out from the respective tables.
The bass boost circuit 170, which is in the form of a shelving filter, performs a filter process, in accordance with the filter coefficient output from the coefficient calculation section 150, for increasing/decreasing a level of a bass range of the input sound signal lower than a predetermined frequency. Similarly, the treble boost circuit 180, which is also in the form of a shelving filter, performs a filter process, in accordance with the filter coefficient output from the coefficient calculation section 160, for increasing/decreasing a level of a treble range of the input sound signal higher than a predetermined frequency.
Because the bass components, extracted from the input sound signal, are boosted by the bass boost circuit 170 as set forth above, the fifth embodiment allows the sound volume of the bass range in the overall sound volume to approach a predetermined volume level and can thereby impart the sound with punch even when the sound source has a small quantity of bass components. Similarly, because the treble components, extracted from the input sound signal, are boosted by the treble boost circuit 180, the instant embodiment can impart the sound with modulation even when the sound source has a small quantity of treble components.
Further, because the fifth embodiment employs feed-forward arrangements for changing in real time the filter coefficients of the bass and treble boost circuits 170 and 180 in accordance with the levels of the bass and treble components extracted by the LPF 110 and HPF 1-20 instead of employing feedback of the level of the sound signal having been subjected to the sound quality adjustment, it can rapidly respond to a sound signal level variation, as compared to the conventionally-known sound quality adjustment device shown in
Further, whereas the fifth embodiment has been described above only in relation to a sound signal of one channel, it may be applied to sound signals of multiple channels. In such a case, the sound quality adjustment device shown in
The sound quality adjustment may be performed separately for each of the channels, in which case same or common gain coefficients are used for these channels. In the case where common gain coefficients are used for the channels, the coefficient calculation section 150 in the bass boost circuit 170 of each of the sound quality adjustment devices for the channels Lch, Rch and Cch may calculate a coefficient corresponding to the greatest level among bass components of three sound signals of the channels Lch, Rch and Cch extracted by the corresponding LPF 110. Thus, in the case where the sound quality adjustment is to be performed for the three channels Lch, Rch and Cch, the same coefficient calculation section can be shared among the respective (i.e., three) bass boost circuits 170 although the sound quality adjustment device has to be provided for each of the channels Lch, Rch and Cch; in this way, the overall circuitry size can be reduced.
The present invention arranged in the above-described manner can be suitably applied to audio apparatus and TV receivers.
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