KR101651828B1 - System and method for improving sound quality of digital sound source - Google Patents
System and method for improving sound quality of digital sound source Download PDFInfo
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- KR101651828B1 KR101651828B1 KR1020150132463A KR20150132463A KR101651828B1 KR 101651828 B1 KR101651828 B1 KR 101651828B1 KR 1020150132463 A KR1020150132463 A KR 1020150132463A KR 20150132463 A KR20150132463 A KR 20150132463A KR 101651828 B1 KR101651828 B1 KR 101651828B1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/002—Damping circuit arrangements for transducers, e.g. motional feedback circuits
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
- G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
- G10H2250/471—General musical sound synthesis principles, i.e. sound category-independent synthesis methods
- G10H2250/475—FM synthesis, i.e. altering the timbre of simple waveforms by frequency modulating them with frequencies also in the audio range, resulting in different-sounding tones exhibiting more complex waveforms
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
- G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
- G10H2250/621—Waveform interpolation
- G10H2250/625—Interwave interpolation, i.e. interpolating between two different waveforms, e.g. timbre or pitch or giving one waveform the shape of another while preserving its frequency or vice versa
Abstract
The digital sound source quality improvement system includes a filter unit, a sound effect adding unit, a rear curve applying unit, and a rear curve correcting unit. The filter unit passes the input sound source through an input filter having a frequency pass characteristic set according to a preset target frequency characteristic, and the sound effect adding unit adds a sound volume corresponding to a frequency that causes a sound source that has passed through the input filter, The rear curve application section adjusts the volume according to the recording characteristics of the RIA curve on the sound source to which the sound effect is added, and the rear curve correction section uses the parallel filter to correct the volume of the sound source controlled by the volume.
Description
BACKGROUND OF THE INVENTION 1. Field of the Invention [0002] The present invention relates to a sound quality improvement system and method, and more particularly, to a system and a method for improving sound quality of a digital sound source such as MP3 or Internet streaming compressed music files.
With the advent of the digital age, most sound sources are now being created and consumed in digital form. However, a digital sound source can not deliver the original analog sound as it is, because it is inevitably in the process of generating it, and also has a selective loss for storage and transmission.
In order to solve this problem, various attempts have been made to improve the sound quality of a digital sound source, such as improvement of a compression method of a digital sound source. However, the conventional attempts have been limited to only some types of compatible digital sound sources, It is difficult to expect effects on other types of digital sound sources that already exist.
SUMMARY OF THE INVENTION It is an object of the present invention to provide a system and method capable of improving sound quality of various types of digital sound sources regardless of the generation or storage method.
In order to achieve the above object, a digital sound source quality improvement system according to the present invention includes an input filter unit, a sound effect adding unit, a rear curve applying unit, and a rear curve correcting unit.
The input filter unit passes an input sound source through an input filter having a frequency pass characteristic set according to a preset target frequency characteristic, and the sound effect adding unit adds a frequency per frequency that causes a sound effect that has been set in advance to the sound source that has passed through the input filter , The Lie curve application part adjusts the volume according to the recording characteristics of the RIA curve on the sound source to which the sound effect is added, and the Lie curve correction part uses the parallel filter to correct the volume of the sound source controlled by the volume .
According to this configuration, regardless of the type of the input digital sound source, the volume of each digital sound source is adjusted by using a preset sound volume frequency characteristic and a correction process, so that various types of digital sound sources The sound quality can be improved.
In this case, the digital sound quality improvement system may further include an input correction unit for correcting the sound volume of the sound source passing through the input filter according to the target sound volume frequency characteristic. According to such a configuration, it becomes possible to adjust the sound volume of each sound source to be closer to the preset target sound volume frequency characteristic.
The apparatus may further include a frequency selection amplifying unit for amplifying a sound volume in a frequency region preset for the sound source corrected by the input correcting unit. According to such a configuration, it is possible to selectively amplify the frequency component only in a specific frequency region such as a low-frequency portion or a high-frequency portion.
The apparatus may further include a harmonic component adding unit that adds a predetermined harmonic component to the sound source when the volume of the sound source to which the sound effect is added is smaller than a predetermined reference size. According to this configuration, if there is room for addition of the volume despite the addition of the sound effect, it is possible to further amplify the harmony that may be lacking, thereby creating a richer sound.
The apparatus may further include a filter frequency characteristic setting unit for calculating a statistical frequency characteristic of the digital sound source from the plurality of digital sound sources and setting the target sound volume frequency characteristic according to the statistical frequency characteristic. According to this configuration, it is possible to set the target sound volume frequency characteristic used for improving the quality of the digital sound source from a plurality of distributed digital sound sources.
In addition, the invention in which the system is implemented in the form of a method is also disclosed.
According to the present invention, regardless of the type of an input digital sound source, the volume of each digital sound source is controlled by using a preset sound volume frequency characteristic and a correction process, so that the sound quality of various types of digital sound sources Can be improved.
In addition, it becomes possible to adjust the sound volume of the sound source according to the frequency closer to the preset target sound volume frequency characteristic.
In addition, the frequency component can be selectively amplified only for a specific frequency region such as a low-frequency portion or a treble portion.
In addition, if there is room for additional volume in spite of the addition of a sound effect, it is possible to further amplify the harmony that may be lacking, thereby producing a richer sound.
In addition, it is possible to set the target sound volume frequency characteristic used for improving the quality of the digital sound source from a plurality of distributed digital sound sources.
1 is a schematic block diagram of a digital sound enhancement system according to an embodiment of the present invention;
FIG. 2 is a schematic flowchart of a method for improving sound quality of a digital sound source according to an embodiment of the present invention. FIG.
FIG. 3 and FIG. 4 are diagrams showing frequency characteristics of a compressed digital sound source file in general; FIG.
5 illustrates the frequency pass characteristics of an input filter according to the present invention.
6 and 7 are graphs showing frequency characteristics of a sound source before and after application of an RIAA curve, respectively.
Hereinafter, preferred embodiments of the present invention will be described with reference to the accompanying drawings.
1 is a schematic block diagram of a digital sound enhancement system according to an embodiment of the present invention.
1, the digital sound source
At this time, each component of the digital sound
The
For this purpose, the frequency characteristic setting unit 180 may calculate the statistical frequency characteristic of the digital sound source from a plurality of digital sound sources, and set the target sound volume frequency characteristic according to the calculated statistical frequency characteristic.
The
The frequency
The sound
The harmonic
The rear
The rear
FIG. 2 is a schematic flowchart of a digital sound quality improvement method according to an embodiment of the present invention.
First, when an MP3 or an online streaming sound source is input, a digital sound source is converted into an analog signal using an equalizer (EQ) and a filter (S110).
3 and 4 are views showing frequency characteristics of a compressed digital sound source file in general. In FIGS. 3 and 4, it can be seen that the portion of 800 - 1500 Hz is significantly highlighted.
A digital sound source having such a frequency characteristic is filtered by a filter having a frequency passing characteristic as shown in Fig. 5, and correction of a deficient portion on the graph is performed. 5 is a diagram illustrating a frequency passing characteristic of an input filter according to the present invention.
Next, the non-linear bending compressor smoothes the sound of the bass part and the treble part which are excessively amplified or distorted during compression (S120). The non-linear bending compressor is a device for bending and amplifying only certain bass parts and specific high-frequency parts of a sound source on the horizontal line.
The processing method of the non-linear bending compressor is similar to the parallel filter, but there are differences. The parallel filter is a kind of mirroring method that adjusts all the values of the same part, whereas the non-linear bending compressor is a method of obtaining a desired value by bending a straight line by a specific part. Here, it is mainly used to control the bass part and the treble part. Only processed to respond to the value obtained by filtering.
Then, the saturation and density are used to fill in the transparency and texture of the cut portion by the filter or the like (S130). Saturation and Density is a device that adjusts the transparency and texture of sound after the first and second machining is completed by adjusting the saturation and density of the sound source to fill the texture of the sound source.
Next, the velocity of the sound source is determined (S140). If it is less than 60, the magnetic EQ and the harmonic distortion process are performed (S142). Magnetic EQ and harmonic distortion are used to amplify the lack of harmonies to create rich sound. Since the velocity is less than 60, a separate amplification process is required. Third and odd harmonics amplification and distortion are added to obtain a coarse and warm analog value.
Magnetic EQ and harmonic distortion have a unique texture that passes through iron, steel, cobalt, and nickel, which is characterized by a different magnetic force. The second, third, and odd harmonies are applied to the device, (More or less) to make the listener unable to think of the loss of sound.
When the velocity of the sound source is 60 or more, RIAA curved equalization is performed (S150). Using RIAA Curved Equalization, apply RIAA Curve to make sound (sound) smoother and analogue in the same way as in Vinyl Mastering.
6 and 7 are views showing frequency characteristics of the sound source before and after applying the RIAA curve, respectively. As can be seen from the graphs of FIGS. 6 and 7, the 1/12 Octave resolution is smoothly changed. In particular, you can see that the high-frequency graph is smoother, and the overall smoothness can be seen, as well as the change in bass, 200 Hz. This method is mainly used for mastering, and it can be obtained a smooth and stable value by applying it.
It is determined whether the velocity of the sound source is less than 50 in step S160. If it is less than 50, the sound is made up in parallel filter using a parallel filter in step S170. A parallel filter is a device used to fill a deficit on the original sound while leaving the original sound intact.
Parallel filters process the same value more accurately than non-linear bending compressors, and can process both the original value and the alias to achieve both amplification and naturalness.
Next, the Multiband Compressor is used to amplify the sound by performing individual amplification functions on a frequency-by-frequency basis. The Multiband Compressor is a device that amplifies parts of the sound that are lacking in three to five parts: bass, mid, treble, bass, bass, mid, middle, and treble.
Finally, normalize using the limiter (S190). The limiter is a device that calculates the amount of sound suitable for final output and calculates and outputs it. Finally, the limiter is necessary for the final stage because it plays the role of making the sound bigger without any shortage.
As a result, the output sound is output as a high quality sound (or output as a more analogized sound) after the loss correction. Therefore, it is possible to output a sound source supplementing the disadvantages of digital.
Although the present invention has been described in terms of some preferred embodiments, the scope of the present invention should not be limited thereby but should be modified and improved in accordance with the above-described embodiments.
100: Digital sound source quality improvement system
110: Input filter unit
120: sound effect adding unit
130: Leica curve application part
140: Lea curve correction unit
150:
160: Frequency selection amplifier section
170: Harmonic component adding unit
Claims (10)
A sound effect adding unit for adding a sound volume for each frequency that causes a sound source that has passed through the input filter to a preset sound effect;
A Leia curve application unit for adjusting the volume of each frequency according to a recording characteristic of a RIA curve in a sound source to which the sound effect is added;
A rear curve correcting unit for correcting the loudness of each frequency of the sound source by using the parallel filter; And
And a harmonic component adding unit for adding a predetermined harmonic component to the sound source when the volume of the sound source to which the sound effect is added is smaller than a preset reference size.
Further comprising an input correcting unit for correcting the sound volume of the sound source passing through the input filter unit according to the target sound volume frequency characteristic.
Further comprising a frequency selection amplifying unit for amplifying a sound volume in a frequency region preset for the sound source corrected by the input correcting unit.
Further comprising a filter frequency characteristic setting unit for calculating a statistical frequency characteristic of a digital sound source from a plurality of digital sound sources and setting the target sound volume frequency characteristic according to the statistical frequency characteristic.
An input filtering step of allowing an input sound source to pass through an input filter having a frequency pass characteristic according to a preset target frequency characteristic;
A sound effect adding step of adding a sound volume for each frequency which causes a sound source that has passed through the input filter to a predetermined sound effect;
Applying a Leia curve to the sound source to which the sound effect is added, the volume of which is adjusted according to the recording characteristics of the RIA curve;
A rear curve correction step of correcting the volume of each of the frequency-adjusted sound sources by using a parallel filter; And
And a harmonic component adding step of adding a preset harmonic component to the sound source when the volume of the sound source to which the sound effect is added is smaller than a preset reference size.
Further comprising an input correcting step of correcting the loudness of the frequency of the sound source that has passed through the filtering step according to the target sound frequency frequency characteristic.
Further comprising a frequency selection amplifying step of amplifying a volume of a frequency region preset for the sound source corrected in the input correcting step.
Further comprising a filter frequency characteristic setting step of calculating a statistical frequency characteristic of a digital sound source from a plurality of digital sound sources and setting the target sound volume frequency characteristic according to the statistical frequency characteristic.
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Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN112585868A (en) * | 2018-06-22 | 2021-03-30 | 杜比实验室特许公司 | Audio enhancement in response to compression feedback |
Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JPH1070795A (en) * | 1996-08-27 | 1998-03-10 | Kawai Musical Instr Mfg Co Ltd | Sound image controller |
KR19990007695A (en) * | 1998-10-22 | 1999-01-25 | 은현기 | Sound signal improving device |
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Patent Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JPH1070795A (en) * | 1996-08-27 | 1998-03-10 | Kawai Musical Instr Mfg Co Ltd | Sound image controller |
KR19990007695A (en) * | 1998-10-22 | 1999-01-25 | 은현기 | Sound signal improving device |
Cited By (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN112585868A (en) * | 2018-06-22 | 2021-03-30 | 杜比实验室特许公司 | Audio enhancement in response to compression feedback |
US11736081B2 (en) * | 2018-06-22 | 2023-08-22 | Dolby Laboratories Licensing Corporation | Audio enhancement in response to compression feedback |
CN112585868B (en) * | 2018-06-22 | 2023-12-05 | 杜比实验室特许公司 | Audio enhancement in response to compressed feedback |
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