Method of converting resolution of video signals and apparatus using the same

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First Claim
1. A method of converting resolution of an input video signal, the method comprising:
 calculating upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
calculating a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes;
calculating first filter coefficients of a same number of filter tabs by multiplying a window function by a sinc function;
calculating final filter coefficients of a filter by subtracting a result of a multiplication of a Gaussian function by the window function from the first filter coefficients, and then normalizing the final filter coefficients; and
performing filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of the input video signal depending on the upsampling and downsampling ratios.
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Abstract
A method converts a resolution of video signals, the method including: calculating upsampling and downsampling ratios based on a resolution of an input video signal and a desired resolution of an output video signal; calculating a number of filter tabs by multiplying the upsampling and downsampling ratios by a number of side lobes; calculating first filter coefficients of a same number of the filter tabs by multiplying a window function by a sinc function; calculating final filter coefficients by subtracting a result of a multiplication of a Gaussian function by a window function from the first filter coefficients, and then normalizing the final filter coefficients; and performing filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of an input video signal depending on the upsampling and downsampling ratios, to obtain clear video images.
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31 Claims
 1. A method of converting resolution of an input video signal, the method comprising:
 calculating upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
calculating a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes;
calculating first filter coefficients of a same number of filter tabs by multiplying a window function by a sinc function;
calculating final filter coefficients of a filter by subtracting a result of a multiplication of a Gaussian function by the window function from the first filter coefficients, and then normalizing the final filter coefficients; and
performing filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of the input video signal depending on the upsampling and downsampling ratios.  View Dependent Claims (2, 3, 4, 5, 6)
 calculating upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
 7. An apparatus to convert resolution of an input video signal, the apparatus comprising:
 a first unit to calculate upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
a second unit to calculate a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes;
a third unit to calculate first filter coefficients of a same number of the filter tabs by multiplying a window function by a sinc function;
a fourth unit to calculate final filter coefficients of a filter by subtracting a result of a multiplication of a Gaussian function by a window function from the first filter coefficients, and then normalizing the final filter coefficients; and
first and second scaling filters to perform filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of the input video signal depending on the upsampling and downsampling ratios.
 a first unit to calculate upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
 8. A computer readable medium having recorded thereon a computer readable program having computerexecutable instructions for converting resolution of an input video signal, the computer instructions comprising:
 calculating upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
calculating a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes;
calculating first filter coefficients of a same number of filter tabs by multiplying a window function by a sinc function;
calculating final filter coefficients of a filter by subtracting a result of a multiplication of a Gaussian function by the window function from the first filter coefficients, and then normalizing the final filter coefficients; and
performing filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of the input video signal depending on the upsampling and downsampling ratios.  View Dependent Claims (9, 10, 11, 12, 13)
 calculating upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal;
 14. An apparatus to convert resolution of an input video signal, the apparatus comprising:
 an input signal processing unit to divide the input video signal into vertical and horizontal direction components;
first and second multiplexers to apply the vertical and horizontal direction components to first and second scaling filters;
the first and second scaling filters to scale the vertical and horizontal direction components according to a predetermined scheme;
a third multiplexer to combine an output of the first and second scaling filters to provide a scaled output to an output signal processing unit; and
the output signal processing unit to convert the scaled output to a desired resolution.  View Dependent Claims (15, 16, 17, 18, 19, 20, 21, 22)
 an input signal processing unit to divide the input video signal into vertical and horizontal direction components;
 23. An apparatus to convert resolution of an input video signal, the apparatus comprising:
 a video signal resolution processing unit to divide the input video signal into vertical and horizontal direction components; and
a bilevel filtering system to perform filtering in vertical and horizontal directions based on final filter coefficients by modifying a sampling rate of the input video signal depending on upsampling and downsampling ratios.  View Dependent Claims (24, 25, 26, 27, 28, 29, 30, 31)
 a video signal resolution processing unit to divide the input video signal into vertical and horizontal direction components; and
1 Specification
This application claims the priority of Korean Patent Application No. 200383612, filed on Nov. 24, 2003, in the Korean Intellectual Property Office, the disclosure of which is incorporated herein in its entirety by reference.
BACKGROUND OF THE INVENTION1. Field of the Invention
The present invention relates to a method of converting input video signals to have a desired resolution and an apparatus using the same, and more particularly, to a method of converting a resolution of video signals, by which the details of the input video may be provided in high definition without additional definition enhancement circuits, such as a peaking circuit, by filtering video signals with optimal filter coefficients being calculated based on a resolution of input and output video signals, and an apparatus using the same.
2. Description of the Related Art
Since digital display devices, such as a liquid crystal display (LCD), a digital micromirror device (DMD), and a plasma display panel (PDP), have a display resolution fixed for each product model, a video input to the individual digital display device has different resolutions, and thus, should be converted to have a resolution adjusted to a corresponding display device.
Particularly, a resolution conversion technique is required to convert a variety of digital television formats, defined by the advanced television system committee (ATSC), into a format which may be reproduced in a high definition television (HDTV).
The resolution conversion techniques allow for the converting of a sampling rate of an input video signal, and the conversion techniques are classified into a resolution extension to convert a lowresolution format into a high definition format and a resolution reduction to convert a highresolution format into a lowresolution format. In the case of the resolution extension, since new signal components are interpolated between samples of an original input signal, a blurring takes place due to losses of high frequency components when the signals are being filtered. Therefore, users can readily recognize a deterioration of display quality when standard definition (SD) video signals are reproduced on a high definition digital display device, such as the HDTV.
Also, in the case of the resolution reduction, since high frequency components in an input video are aliased on a low frequency signal, deterioration, such as a zigzag artifact and a moiré pattern, occurs.
According to conventional linear filtering techniques as disclosed in U.S. Pat. No. 5,889,895 and U.S. Pat. No. 5,671,298, resolution conversion is accomplished by using a bilinear interpolation and a cubic interpolation. However, since high frequency components of the input video are not sufficiently extended during the resolution extension, both the definition and the display quality deteriorate. To compensate for such a problem, a method was proposed in which a peaking is applied to a low resolution video to identify potential edge pixels, and then edge pixel detection, edge linking, and luminance transition enhancement are sequentially accomplished so as to output high definition video signals. However, such a scaling method uses a conventional linear filter, and thus, has a problem that both preprocessing and postprocessing require an increase in arithmetic operations and additional hardware, thus causing costs to increase because the peaking and the luminance transition enhancement should be accomplished for the video signals in both preprocessing and postprocessing stages during filtering to improve the display quality and the definition of a video.
In addition, according to the conventional art disclosed in U.S. Pat. No. 5,852,470 and U.S. Pat. No. 5,446,804, video signals corresponding to the edge regions are processed satisfactorily. However, fine textured regions of a video cannot be processed with high definition. In addition, their performances are unsatisfactory compared to the linear filtering technique, with the exception of most regions of the edge components.
SUMMARY OF THE INVENTIONThe present invention provides a method of converting a resolution, to reproduce clearly an input video with a desired resolution with neither preprocessing nor postprocessing, such as a peaking or a luminance transition enhancement during a resolution conversion process, by calculating optimal filter coefficients based on each resolution of input and output video signals and applying the coefficients to a scaling filter, and an apparatus using the same.
According to an aspect of the present invention, a method converts the resolution of video signals, the method comprising: calculating upsampling and downsampling ratios based on the resolution of an input video signal and the desired resolution of an output video signal; calculating a number of filter tabs by multiplying the upsampling and downsampling ratios by a number of side lobes; calculating first filter coefficients of the same number of the filter tabs by multiplying a window function by a sinc function; calculating final filter coefficients by subtracting a result of a multiplication of a Gaussian function by the window function from the first filter coefficients, and then normalizing the final filter coefficients; and performing filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of an input video signal depending on the upsampling and downsampling ratios.
The upsampling and downsampling ratios may be calculated by using a greatest common measure of both a number of samples of the input video signal and a number of samples of a video signal having a desired definition.
The number of filter tabs may be calculated by using an equation:<FORM>T=round(max{U,D}×SmoothingAmount×(nLobes−1))×2+1, </FORM>where T is the number of filter tabs, nLobes is the number of side lobes, U and D are optimal upsampling and downsampling ratios, and SmoothingAmount is a constant for modifying a cutoff frequency of the filter.
The value of SmoothingAmount may be set to be less than 1, and the value of nLobes may be set to be less than 2.
The first filter coefficients may be calculated by using an equation:<maths id="MATHUS00001" num="1"><math overflow="scroll"><mrow><mrow><mrow><mi>h</mi><mo></mo><mrow><mo>[</mo><mi>i</mi><mo>]</mo></mrow></mrow><mo>=</mo><mrow><mrow><mo>{</mo><mfrac><mrow><mi>sin</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow><mi>x</mi></mfrac><mo>}</mo></mrow><mo>×</mo><mrow><mi>Kaiser</mi><mo></mo><mrow><mo>(</mo><mrow><mi>i</mi><mo>,</mo><mi>β</mi></mrow><mo>)</mo></mrow></mrow></mrow></mrow><mo>,</mo><mrow><mi>i</mi><mo>=</mo><mn>0</mn></mrow><mo>,</mo><mn>1</mn><mo>,</mo><mi>…</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo>,</mo><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mo>,</mo><mstyle><mtext></mtext></mstyle><mo></mo><mrow><mi>x</mi><mo>=</mo><mrow><mfrac><mrow><mi>i</mi><mo></mo><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mrow><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mfrac><mo>×</mo><mi>π</mi><mo>×</mo><mi>nLobes</mi></mrow></mrow></mrow></math></maths>where, sin(x)/x is an ideal low frequency band pass function, and Kaiser(I, β) is a Kaiser window function.
The final filter coefficients may be defined as:<maths id="MATHUS00002" num="2"><math overflow="scroll"><mrow><mrow><mrow><mi>h</mi><mo></mo><mrow><mo>[</mo><mi>i</mi><mo>]</mo></mrow></mrow><mo>=</mo><mrow><mrow><mo>[</mo><mrow><mfrac><mrow><mi>sin</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow><mi>x</mi></mfrac><mo></mo><mrow><mi>ES</mi><mo>·</mo><mrow><mi>Gaussian</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow></mrow></mrow><mo>]</mo></mrow><mo>·</mo><mrow><mi>Kaiser</mi><mo></mo><mrow><mo>(</mo><mrow><mi>i</mi><mo>,</mo><mi>β</mi></mrow><mo>)</mo></mrow></mrow></mrow></mrow><mo>,</mo><mstyle><mtext></mtext></mstyle><mo></mo><mrow><mi>i</mi><mo>=</mo><mn>0</mn></mrow><mo>,</mo><mn>1</mn><mo>,</mo><mi>…</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo>,</mo><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mo>,</mo><mrow><mi>x</mi><mo>=</mo><mrow><mfrac><mrow><mi>i</mi><mo></mo><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mrow><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mfrac><mo>×</mo><mi>π</mi><mo>×</mo><mi>nLobes</mi></mrow></mrow></mrow></math></maths>where ES is a parameter for determining a magnitude of a high frequency signal in a pass band, and Kaiser(I, β) is a Kaiser window function.
According to another embodiment of the present invention, an apparatus converts resolution of video signals, the apparatus comprising: a unit to calculate upsampling and downsampling ratios based on a resolution of an input video signal and a desired resolution of an output video signal; a unit to calculate a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes; a unit to calculate first filter coefficients of the same number of the filter tabs by multiplying a window function by a sinc function; a unit to calculate final filter coefficients by subtracting a result of a multiplication of a Gaussian function by a window function from the first filter coefficients, and then normalizing the final filter coefficients; and first and second scaling filters to perform filtering in vertical and horizontal directions, respectively, based on the final filter coefficients by modifying a sampling rate of an input video signal depending on the upsampling and downsampling ratios.
Additional aspects and/or advantages of the invention will be set forth in part in the description which follows and, in part, will be obvious from the description, or may be learned by practice of the invention.
BRIEF DESCRIPTION OF THE DRAWINGSThese and/or other aspects and advantages of the invention will become apparent and more readily appreciated from the following description of the embodiments, taken in conjunction with the accompanying drawings of which:
FIG. 1 is a block diagram illustrating a sampling rate conversion for input video signals according to an embodiment of the present invention;
FIG. 2 is a block diagram illustrating an Lfold upsampler in FIG. 1;
FIG. 3A illustrates a spectrum of an input signal x(n) in FIG. 2;
FIG. 3B illustrates a spectrum in which samples of the input signal x(m) in FIG. 2 are upsampled by L;
FIG. 4 is a block diagram illustrating a first low pass filter connected to the Lfold upsampler in FIG. 2;
FIG. 5 is a block diagram illustrating an Mfold downsampler in FIG. 1;
FIG. 6A illustrates a spectrum of an input signal x<sub>2</sub>(n) in FIG. 5;
FIG. 6B illustrates a spectrum in which samples of the input signal x<sub>2</sub>(n) in FIG. 5 are downsampled by M;
FIG. 7 is a block diagram illustrating a second low pass filter connected to the Mfold downsampler in FIG. 5;
FIG. 8 is a block diagram illustrating integrating the upsampling and downsampling processing units in FIGS. 4 and 7;
FIG. 9 is a schematic diagram showing a cutoff frequency, a transition bandwidth, and an amount of stop band attenuation considered during filter designing in an embodiment of the present invention;
FIGS. 10A, 10B, and 10C are graphs of frequency responses;
FIG. 11 is a graph showing a frequency response of a low pass filter having a cutoff frequency of 500 Hz according to an embodiment of the present invention; and
FIG. 12 is a block diagram illustrating a typical resolution conversion unit of a method of converting resolution according to an embodiment of the present invention.
DETAILED DESCRIPTION OF THE EMBODIMENTSReference will now be made in detail to the embodiments of the present invention, examples of which are illustrated in the accompanying drawings, wherein like reference numerals refer to the like elements throughout. The embodiments are described below to explain the present invention by referring to the figures.
FIG. 1 is a block diagram illustrating a sampling rate conversion for input video signals according to an embodiment of the present invention. According to a typical sampling conversion technique, the upsampler 10 performs a zero insertion between the pixels of an input video signal x(n) and outputs an upsampled video signal. The interpolation filter 20 performs low pass filtering for the upsampled video signal based on a received filter coefficient, and the downsampler 30 performs downsampling for the filtered video signal suitably for a desired resolution.
FIG. 2 is a block diagram illustrating an Lfold upsampler in FIG. 1. The Lfold upsampler inserts (L−1) zeros between the samples of the Lfold upsampled input video signal. Supposing that samples of the input signal are x(n)={ . . . , 3, 5, 9, 6, . . . } and L=4, samples of the output signal x<sub>1</sub>(n) become x<sub>1</sub>(n)={ . . . , 3, 0, 0, 0, 5, 0, 0, 0, 9, 0, 0, 0, 6, 0, 0, 0, . . . }, that is, three zeros are inserted between each of the samples of the input signal x(n). This can be represented by the following general expression.<FORM>x<sub>1</sub>,(n)=[L]X(n)=x(n/L); when n is a multiple of L =0 ; otherwise Equation 1 </FORM>
FIG. 3A illustrates a spectrum of an input signal x(n) in FIG. 2. FIG. 3B illustrates a spectrum in which samples of the input signal x(n) in FIG. 2 are upsampled by L. FIG. 4 is a block diagram illustrating a first low pass filter 22 connected to the Lfold upsampler 10 in FIG. 2.
The spectrum of the input signal x(n) is compressed into (L1) spectral components with a range of −π˜+π. The first low pass filter 22 has a cutoff frequency set at π/L to pass only the spectral components 32 positioned in a low frequency band in the spectrum of the compressed input signal x<sub>1</sub>(n) as shown in FIG. 3B.
FIG. 5 is a block diagram illustrating an Mfold downsampler 30. The Mfold downsampler is a circuit from which an input signal is output without being altered only when the position of the input sample is an integer multiple of M. Supposing that the samples of the input signal x<sub>2</sub>(n) are x<sub>2</sub>(n)={ . . . , 7, 3, 5, 2, 9, 6, 4, . . . }, M=2, and x<sub>2</sub>(0)=5, the samples of the output signal become x<sub>3</sub>(n)={ . . . , 7, 5, 9, 4, . . . }. This can be represented as the following general expression.<FORM>x<sub>3</sub>(n)=[↓M]X<sub>2</sub>(n)=x<sub>2</sub>(n/M), when n is a multiple of M=0, otherwise Equation 2 </FORM>
FIG. 6A illustrates a spectrum of an input signal x<sub>2</sub>(n) in FIG. 5. FIG. 6B illustrates a spectrum in which samples of the input signal x<sub>2</sub>(n) in FIG. 5 are downsampled by M. FIG. 7 is a block diagram illustrating a second low pass filter 24 connected to the Mfold downsampler in FIG. 5. Frequency bandwidths of the spectra shown in FIGS. 6A and 6B have been adjusted to give a convenient description.
The spectrum of the Mfold downsampled signal x<sub>3</sub>(n) is formed by extending the input signal x<sub>2</sub>(n) by X[M] to have (M−1) spectral components, so that an aliasing occurs due to a superposition with the spectrum of the input signal x<sub>2</sub>(n). To prevent such an aliasing, the input signal x<sub>2</sub>(n) is passed through the second low pass filter 24 having its cutoff frequency of π/M before downsampling as shown in FIG. 7. In addition, to remove additional spectral components caused by the upsampling and to prevent the aliasing caused by the downsampling, the interpolation filter 20 in FIG. 1 is set to have the lowest cutoff frequency, min(π/L, π/M), of the first and second low pass filters 22 and 24.
There are a variety of methods of evaluating a filter coefficient of the interpolation filter 20, a finite impulse response (FIR) filter. According to the present invention, a windowbased design method is adopted due to the convenience of controlling the amount of stop band attenuation and a transition bandwidth, which is important for determining a filter property.
FIG. 9 is a schematic diagram showing a cutoff frequency, a transition bandwidth, and an amount of stop band attenuation considered during filter designing in an embodiment of the present invention. During the filter designing, a narrow transition bandwidth and a large amount of stop band attenuation may prevent the video quality deterioration, such as ringing and aliasing which are caused by the filtering. The type of the window function determines the frequency characteristics of the filter. That is, as the width of a main lobe of the window function becomes narrower, the designed filter has a larger amount of stop band attenuation.
In the fields of the filter designing, various kinds of window functions are being adopted to optimize the transition bandwidth and the amount of stop band attenuation. According to the present invention, a Kaiser window function is adopted because the bandwidth of the main lobe and the ripple of the side lobe of the window function may be conveniently controlled.
An impulse response h(n) of a typical window function may be expressed as the following equation.<FORM>h(n)=h<sub>d</sub>(n)×w(n) Equation 3 </FORM>where, h<sub>d</sub>(n) is an impulse response of an ideal low pass filter, and w(n) is a window function.
The window function w(n) may be expressed as the following equation, the Kaiser window function.<maths id="MATHUS00003" num="3"><math overflow="scroll"><mtable><mtr><mtd><mtable><mtr><mtd><mrow><mrow><mrow><mi>Kaiser</mi><mo></mo><mrow><mo>(</mo><mrow><mi>n</mi><mo>,</mo><mi>β</mi></mrow><mo>)</mo></mrow></mrow><mo>=</mo><mi/><mo></mo><mfrac><mrow><msub><mi>I</mi><mn>0</mn></msub><mo>[</mo><mrow><mi>β</mi><mo>(</mo><mrow><mn>1</mn><mo></mo><msup><mrow><mo>[</mo><mrow><mrow><mo>(</mo><mrow><mi>n</mi><mo></mo><mi>α</mi></mrow><mo>)</mo></mrow><mo>/</mo><msup><mi>α</mi><mn>2</mn></msup></mrow><mo>]</mo></mrow><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msup></mrow><mo>]</mo></mrow></mrow><mrow><msub><mi>I</mi><mn>0</mn></msub><mo></mo><mrow><mo>(</mo><mi>β</mi><mo>)</mo></mrow></mrow></mfrac></mrow><mo>,</mo></mrow></mtd><mtd><mrow><mi/><mo></mo><mrow><mrow><mn>0</mn><mo>≤</mo><mi>n</mi><mo>≤</mo><mi>T</mi></mrow><mo>,</mo><mrow><mi>α</mi><mo>=</mo><mrow><mi>T</mi><mo>/</mo><mn>2</mn></mrow></mrow></mrow></mrow></mtd></mtr><mtr><mtd><mrow><mrow><mo>=</mo><mi/><mo></mo><mn>0</mn></mrow><mo>,</mo></mrow></mtd><mtd><mrow><mi/><mo></mo><mi>otherwise</mi></mrow></mtd></mtr></mtable></mtd><mtd><mrow><mi>Equation</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo></mo><mn>4</mn></mrow></mtd></mtr></mtable></math></maths>where T is the number of filter tabs, I<sub>0 </sub>is a modified zeroorder Bessel function, and α and β are coefficients to determine a configuration of a Kaiser window. The frequency characteristics of the Kaiser window function are determined by coefficients β and T. As the β increases, the stop band attenuation decreases. As the T increases, the main lobe of the window function becomes narrower. Therefore, the transition bandwidth is reduced.
Ideally, the interpolation filter 20 to convert a resolution should have a frequency response which is flat in the pass band and which has a larger amount of attenuation in the stop band to prevent aliasing. Particularly, in the multiples of the sampling frequency, the interpolation filter 20 generally has a very high stop band attenuation to prevent aliasing in direct current (DC) components of the input signal because the aliasing may be recognized accurately by the naked eye. In addition, to prevent ringing and overshooting in the edge regions of images, it is recommended that the impulse response of the interpolation filter 20 have a smaller number of side lobe components and smaller side lobes.
According to the present invention, when determining the filter coefficients, a filter tab number (T) is calculated by using the stop band attenuation and the transition region bandwidth, which are not in a tradeoff, as is shown in the following equation.<FORM>T=round(max{U,D}×SmoothingAmount×(nLobes−1))×2+1 Equation 5 </FORM>where round is a rounding function, nLobes is the number of side lobes in the impulse response, and U and D are optimal upsampling and downsampling ratios, respectively. A greatest common measure of both the number of samples in the input signal and the number of samples in the output signal is calculated, and then the numbers of the samples in the input signal and the output signal are each divided by the greatest common measure to obtain optimal upsampling and downsampling ratios, respectively. The optimized upsampling and downsampling ratios are used to determine the cutoff frequency of the filter. Typically, the number of side lobes in the impulse response is directly proportional to the number of the filter tabs (T). The number of filter tabs (T) may be calculated by multiplying the number of side lobes (nLobes) by the upsampling and downsampling ratios. Herein, SmoothingAmounting is a parameter for modifying the cutoff frequency of the filter, and becomes directly proportional to the number of filter tabs and the cutoff frequency if the number of side lobes is determined. It is for this reason that the equation to calculate the number of filter tabs includes the parameter, SmoothingAmounting.
Generally, SmoothingAmounting is set to be less than 1, and nLobes is set to be less than 2 in the Equation 5. A filter coefficient h[i] may be obtained from the following equation.<maths id="MATHUS00004" num="4"><math overflow="scroll"><mtable><mtr><mtd><mrow><mrow><mrow><mi>h</mi><mo></mo><mrow><mo>[</mo><mi>i</mi><mo>]</mo></mrow></mrow><mo>=</mo><mrow><mrow><mo>{</mo><mfrac><mrow><mi>sin</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow><mi>x</mi></mfrac><mo>}</mo></mrow><mo>×</mo><mrow><mi>Kaiser</mi><mo></mo><mrow><mo>(</mo><mrow><mi>i</mi><mo>,</mo><mi>β</mi></mrow><mo>)</mo></mrow></mrow></mrow></mrow><mo>,</mo><mrow><mi>i</mi><mo>=</mo><mn>0</mn></mrow><mo>,</mo><mn>1</mn><mo>,</mo><mi>…</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo>,</mo><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mo>,</mo><mstyle><mtext></mtext></mstyle><mo></mo><mrow><mi>x</mi><mo>=</mo><mrow><mfrac><mrow><mi>i</mi><mo></mo><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mrow><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mfrac><mo>×</mo><mi>π</mi><mo>×</mo><mi>nLobes</mi></mrow></mrow></mrow></mtd><mtd><mrow><mi>Equation</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo></mo><mn>6</mn></mrow></mtd></mtr></mtable></math></maths>where x is a scaling constant factor to allow the sinc function to have the number of side lobes integrated in the Equation 5 within the range of zero and the number of filter tabs (0˜L−1). The filter coefficients calculated by the Equation 6 are normalized to produce a constant output signal for a constant continuous input signal, that is, a flat signal.
Since an interpolation filter is typically used as a method of changing a sampling rate, spectrum attenuation is generated in the high frequency band of the input signal. This causes a degradation of the definition in the filtered video, which would be readily recognized. To compensate for this problem, according to an embodiment of the present invention, the magnitude of the frequency response of the high frequency signal in the pass band of the filter is forced to increase with the number of the filter tabs remaining constant during the generation of the filter coefficients, thus improving the definition. For this purpose, a Gaussian function is subtracted from the original filter kernel in Equation 6 to calculate the filter coefficients as expressed in the following Equation 7. Subsequently, the final filter coefficients are obtained by normalization.<maths id="MATHUS00005" num="5"><math overflow="scroll"><mtable><mtr><mtd><mrow><mrow><mrow><mi>h</mi><mo></mo><mrow><mo>[</mo><mi>i</mi><mo>]</mo></mrow></mrow><mo>=</mo><mrow><mrow><mo>[</mo><mrow><mfrac><mrow><mi>sin</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow><mi>x</mi></mfrac><mo></mo><mrow><mi>ES</mi><mo>·</mo><mrow><mi>Gaussian</mi><mo></mo><mrow><mo>(</mo><mi>x</mi><mo>)</mo></mrow></mrow></mrow></mrow><mo>]</mo></mrow><mo>·</mo><mrow><mi>Kaiser</mi><mo></mo><mrow><mo>(</mo><mrow><mi>i</mi><mo>,</mo><mi>β</mi></mrow><mo>)</mo></mrow></mrow></mrow></mrow><mo>,</mo><mstyle><mtext></mtext></mstyle><mo></mo><mrow><mi>i</mi><mo>=</mo><mn>0</mn></mrow><mo>,</mo><mn>1</mn><mo>,</mo><mi>…</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo>,</mo><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mo>,</mo><mrow><mi>x</mi><mo>=</mo><mrow><mfrac><mrow><mi>i</mi><mo></mo><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mrow><mfrac><mrow><mi>T</mi><mo></mo><mn>1</mn></mrow><mn>2</mn></mfrac></mfrac><mo>×</mo><mi>π</mi><mo>×</mo><mi>nLobes</mi></mrow></mrow></mrow></mtd><mtd><mrow><mi>Equation</mi><mo></mo><mstyle><mtext> </mtext></mstyle><mo></mo><mn>7</mn></mrow></mtd></mtr></mtable></math></maths>where ES is a control factor to determine a magnitude of a high frequency signal in a pass band. Supposing that H(W) is a frequency response calculated by using the filter coefficient obtained from Equation 6, and G(W) is a frequency response, Gaussian(x). Kaiser(i, β) of the Gaussian filter in Equation 7, the final frequency response of the filter generated from Equation 7 may be expressed as H(W)−ES×G(W). Herein, as the gain ES of the high frequency signal becomes smaller, the final frequency response becomes closer to the original frequency response H(W) of the filter. In addition, as the control factor ES increases, the magnitude response gain in the low frequency band decreases. Such a smaller magnitude response may be compensated for by normalizing the filter coefficient.
FIG. 10A shows a graph of a frequency response when filtering is performed by using the interpolation filter in FIG. 1 and the filter coefficients in Equation 6 according to an embodiment of the present invention. Herein, A is a frequency response, and B is a frequency response of a Gaussian function.
FIG. 10B shows a graph of a frequency response when filtering is performed by using the interpolation filter in FIG. 1 and the filter coefficient in Equation 7.
FIG. 10C shows a graph of a frequency response when the filter coefficients applied in FIG. 10B are normalized, and then filtering is performed. It is recognized that the high frequency components in the input signal may be effectively emphasized without modifying the number of filter tabs.
FIG. 11 is a graph showing a frequency response of a low pass filter having a cutoff frequency of 500 Hz according to an embodiment of the present invention. In FIG. 11, A′ is a frequency response of an ideal low pass filter, B′ is a frequency response obtained by using the filter coefficients in Equation 6 according to an embodiment of the present invention, and C is a frequency response of a filter in which the magnitude of high frequency components in the pass band is enhanced based on the filter coefficients obtained from Equation 7.
As shown in FIG. 11, high frequency components in the pass band may be effectively increased without any degradation of the stop band attenuation in the high frequency band. This improves video definition.
Referring to FIG. 12, a block to convert a resolution with respect to the vertical and horizontal directions of the video signals YCbCr and RGB is illustrated. The video signal divided into the vertical and horizontal directions in the input signal processing unit 100 is applied to first and second scaling filters 130 and 140 through first and second multiplexers 110 and 120, respectively, and then converted to have a desired resolution. After the filtering, the signal is output to the output signal processing unit 160 through the third multiplexer unit 150. Herein, each of the first and second scaling filters 130 and 140 is a sampling conversion block including an interpolation filter 20 as shown in FIG. 1 that performs filtering for video signals based on the filter coefficients input from a unit to calculate filter coefficients (not shown in the drawings) so that a high definition video may be provided even after converting a resolution.
According to the present invention, since a resolution of an output video may be freely converted, video of different resolutions may be supported in a variety of digital display devices. Additionally, in spite of the fact that the transition region bandwidth and the stop band attenuation of the interpolation filter are in a tradeoff, the transition region bandwidth and the stop band attenuation of the interpolation filter may be used to calculate optimal filter coefficients and to control the interpolation filter. Therefore, high definition output video signals are provided without adding a peaking circuit or a definition enhancement circuit.
Also, it is possible to control the definition, aliasing and ringing properties of the output video accurately by controlling the control factor ES in the equation that calculates filter coefficients.
The present invention may be embodied as a program stored on a computer readable medium that can be run on a general computer. Here, the computer readable medium includes, but is not limited to, storage media such as magnetic storage media (e.g., ROM's, floppy disks, hard disks, and the like), optically readable media (e.g., CDROMs, DVDs, etc.), and carrier waves (e.g., transmission over the Internet). The present invention may also be embodied as a computer readable program code unit stored on a computer readable medium, for causing a number of computer systems connected via a network to affect distributed processing.
In one embodiment, an apparatus to convert resolution of an input video signal in accordance with the present invention comprises, the apparatus comprises: a video signal resolution processing unit to divide the input video signal into vertical and horizontal direction components; and a bilevel filtering system to perform filtering in vertical and horizontal directions based on final filter coefficients by modifying a sampling rate of the input video signal depending on upsampling and downsampling ratios.
The bilevel filtering system calculates upsampling and downsampling ratios based on a resolution of the input video signal and a desired resolution of an output video signal; calculates a number of filter tabs by multiplying upsampling and downsampling ratios by a number of side lobes; calculates first filter coefficients of a same number of the filter tabs by multiplying a window function by a sinc function; calculates final filter coefficients of a filter by subtracting a result of a multiplication of a Gaussian function by a window function from the first filter coefficients, and then normalizing the final filter coefficients; performs filtering in vertical and horizontal directions based on the final filter coefficients by modifying a sampling rate of an input video signal depending on the upsampling and downsampling ratios; and scales a combined output from the filtering in vertical and horizontal directions and converts the scaled combined output to a desired resolution.
Although a few embodiments of the present invention have been shown and described, it would be appreciated by those skilled in the art that changes may be made in these embodiments without departing from the principles and spirit of the invention, the scope of which is defined in the claims and their equivalents.