JP4760278B2 - Interpolation device, audio playback device, interpolation method, and interpolation program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To improve sound quality by suitably interpolating harmonic components to a high-frequency band. <P>SOLUTION: A divided filter bank 12 generates information by specified frequency bands from audio waveform data upsampled by an upsampler 11. An interpolation processing means 13 judges correlation of a high-frequency side sub-band within a range of a frequency component of the audio waveform data input to the upsampler 11 with sub-bands by the specified frequency bands generated by the divided filter bank 12 and interpolates information by frequencies of correlative sub-bands as high-frequency side frequency information. A composite filter bank 14 generates audio waveform data from the information on the specified frequency bands of the interpolated audio waveform data. <P>COPYRIGHT: (C)2007,JPO&amp;INPIT

Description

  The present invention relates to an interpolation device, an audio playback device, an interpolation method, and an interpolation program.

  Patent Document 1 discloses an audio signal reproduction device. In this audio signal reproducing apparatus, the high frequency sound signal generating means generates a high frequency sound signal, and the signal superimposing means superimposes the high frequency sound signal on the analog audio signal output from the D / A conversion means. . The high-frequency sound signal generated by the high-frequency sound signal generating means has a frequency component exceeding the cut-off frequency of the analog audio signal, and may have a fluctuation component.

JP-A-2-68773 (Claims, Detailed Description of the Invention, etc.)

  Digital recording and playback of music and the like is common. When digitally recording music or the like, a waveform signal of music is sampled at a sampling frequency, and quantized bit data obtained by sampling is recorded.

  CD-DA (Compact Disc Digital Audio) has been used for digital recording of music and the like. In CD-DA, the sampling frequency is 44.1 kHz, and the quantization bit is 16 bits. It is said that the frequency of sounds that humans can hear is about 20 kHz. By setting the sampling frequency to 44.1 kHz, the CD-DA can digitize up to a frequency component of about 22 kHz. It is possible to cover a frequency band (for example, 0 to 20 kHz) of a sound that can be heard by a person.

  However, even if the sound quality covers the sound range that is said to be audible to humans in this way, it may appeal to the reproduced sound of the CD-DA. Some people complain of a sense of incongruity with the playback sound, such as lack of luster or sound. In particular, it seems that there are many people who complain of a sense of incongruity in the reproduced sound when listening to music using many acoustic instruments such as classics and orchestras.

  In recent years, a new music digital recording system has been proposed and put into practical use. For example, SACD (Super Audio CD). In SACD, a DSD (direct stream digital) signal is recorded and reproduced. The sampling frequency of the DSD signal is 2822.4 kHz. Thereby, the frequency component of the sound up to about 80 to 100 kHz can be digitized.

  In this way, when the next generation digital recording system appears and the generation of music digital recording system is changed, the music recorded with the conventional CD-DA becomes data with poor sound quality, and its utility value decreases. There is a possibility that it will.

  Therefore, it is conceivable to improve the sound quality of the music recorded by the conventional CD-DA using the technique described in Patent Document 1. However, in Patent Document 1, a high-frequency sound signal that is not related to the audio signal output from the oversampling digital low-pass filter is superimposed. This high frequency component is generated by the high frequency sound signal generating means, and has a noise-like sound quality from the output audio signal. Even if this noisy high frequency component is added to the original signal as a high frequency component, the sound quality cannot be improved.

  An object of the present invention is to obtain an interpolation device, an audio reproduction device, an interpolation method, and an interpolation program that can suitably interpolate a harmonic component such as a musical instrument in a high frequency band and thereby improve sound quality.

The interpolating apparatus according to the present invention receives input audio waveform data having frequency components in the audible band, and inputs at least an input from an upsampler for increasing the sampling frequency of the audio waveform data and audio waveform data upsampled by the upsampler. A divided filter bank that generates a plurality of subbands for each predetermined frequency band within a range of frequency components of the audio waveform data, and an input audio waveform for the plurality of subbands for each predetermined frequency band generated by the divided filter bank The correlation with the highest frequency subband within the range of the frequency component of the data is determined, and there is a correlation based on the frequency component of the predetermined frequency band of the highest frequency subband and the subband correlated therewith. All of the determined subbands And a band there is one correlation, and interpolation processing means for interpolating the entire band is the correlation, as frequency components above the upper limit of the frequency components of the input audio waveform data, a predetermined frequency band of the subband having a correlation from the frequency components of a predetermined frequency band of the audio waveform data frequency component is interpolated, those having a synthesis filter bank for generating audio waveform data.

  In addition to the configuration of the above-described invention, the interpolating apparatus according to the present invention has audio waveform data input to the upsampler having an upper limit of the frequency component that is not less than 10 kHz lower than the upper limit of the audible band.

In the interpolation device according to the present invention, in addition to the components of the above-described invention, the division filter bank is within the range of the frequency components of the input audio waveform data among the frequency components of the audio waveform data upsampled by the upsampler. Only, a plurality of subbands for each predetermined frequency band are generated.

In addition to the components of the above-described invention, the interpolation apparatus according to the present invention has a low-pass filter that extracts frequency components below the upper limit of the frequency components of the input audio waveform data from the audio waveform data upsampled by the upsampler. The division filter bank generates a plurality of subbands for each predetermined frequency band for the frequency components extracted by the low-pass filter.

The interpolation device according to the present invention, in addition to each configuration of the above-described invention, the division filter bank, for all the frequency components below the Nyquist frequency in the frequency components of the audio waveform data upsampled by the upsampler, A plurality of subbands for each predetermined frequency band are generated .

  In addition to the components of the above-described invention, the interpolation apparatus according to the present invention is improved by the upsampler from the upper limit of the frequency component of the input audio waveform data by the interpolation processing means repeatedly interpolating correlated subbands. It fills up to the Nyquist frequency of the sampled audio waveform data.

  The interpolation apparatus according to the present invention includes, in addition to the components of the above-described invention, intensity distribution determination means for predicting an intensity lower than the intensity of input audio waveform data input to the upsampler as the intensity of the component to be interpolated, And a variable equalizer that changes the intensity of at least the interpolated frequency component among the frequency components of the audio waveform data generated by the synthesis filter bank on the basis of the determined intensity.

  An audio reproducing apparatus according to the present invention includes a decoder that generates audio waveform data having a predetermined sampling frequency having a frequency component in an audible band, and the audio waveform data generated by the decoder as input audio waveform data. An interpolation apparatus according to each configuration of the above-described invention to be interpolated, and an audio amplifier that generates an audio waveform signal from the audio waveform data interpolated by the interpolation apparatus.

The interpolation method according to the present invention includes a step of increasing the sampling frequency of audio waveform data having frequency components in the audible band, and at least within the frequency component range of the input audio waveform data from the increased audio waveform data. of, generating a plurality of sub-bands for each predetermined frequency band, for a plurality of subbands of each predetermined frequency band produced by the filter banks, the highest frequency side within the range of the frequency components of the input audio waveform data Based on the frequency components of the predetermined frequency band of the subband on the highest frequency side and the subband correlated therewith, the whole of the plurality of subbands determined to be correlated is determined as one subband. A band with correlation and the entire band with correlation is input A step of interpolating a frequency component above the upper limit of the frequency components of the I Oh waveform data, from the frequency components of a predetermined frequency band of the audio waveform data frequency components are interpolated in a predetermined frequency band of the subband having a correlation, audio waveform Generating data.

An interpolation program according to the present invention is input to a computer from input audio waveform data having frequency components in an audible band, an upsampler for increasing the sampling frequency of the audio waveform data, and audio waveform data upsampled by the upsampler. , for the range of the frequency components of at least the input audio waveform data, and division filter bank for generating a plurality of sub-bands for each predetermined frequency band, for a plurality of subbands of each predetermined frequency band produced by the filter banks, Determine the correlation with the highest frequency subband within the range of the frequency component of the input audio waveform data, and correlate based on the frequency component of the predetermined frequency band of the highest frequency subband and the subband correlated therewith. Judge that there is The entire plurality of sub-bands, and the band there is one correlation, and interpolation processing means for interpolating the entire band is the correlation, as frequency components above the upper limit of the frequency components of the input audio waveform data, a correlation It is made to function as a synthesis filter bank that generates audio waveform data from the frequency components of the predetermined frequency band of the audio waveform data interpolated with the frequency components of the predetermined frequency band of the subband.

  In the present invention, harmonic components such as musical instruments are preferably interpolated in the high frequency band, thereby improving the sound quality.

  Hereinafter, an interpolation device, an audio reproduction device, an interpolation method, and an interpolation program according to embodiments of the present invention will be described with reference to the drawings. The audio playback apparatus will be described by taking a portable audio player as an example. The interpolation device and the interpolation program will be described as a part of the configuration of the audio playback device. The interpolation method will be described as part of the operation of the audio playback device.

Embodiment 1 FIG.
FIG. 1 is a block diagram showing a portable audio player according to Embodiment 1 of the present invention. The portable audio player includes a hard disk drive (HDD) 1, a decoder 2, an interpolation unit 3 as an interpolation device, an audio amplifier 4, and a headphone jack 5. In the present embodiment, the interpolation unit 3 interpolates an audible out-of-band component.

  The hard disk drive 1 stores lossless compressed data 6 such as music. The hard disk drive 1 stores lossless compressed data 6 of a plurality of music pieces.

  The lossless compressed data 6 is music data that can be reversibly converted to linear PCM (Pulse Code Modulation) data used for music recording on a CD (Compact Disc) or the like, and has a smaller data amount than the linear PCM data. Is.

  Linear PCM data as one type of audio waveform data is data obtained by sampling music and the like at regular intervals. The linear PCM data for CD is data sampled at a sampling frequency of 44.1 kHz. In this case, frequency components up to about 22 kHz Nyquist frequency, which is half the sampling frequency, can be converted into data. The upper limit of the human audible band is said to be about 20 kHz. The linear PCM data for CD covers this general human audible band. A frequency component of 22 kHz or higher, that is, a high frequency component outside the audible band is basically not included. Further, the quantization bit of the linear PCM data for CD is 16 bits.

  The lossless compressed data 6 may be stored in a storage device other than the hard disk drive 1, such as a semiconductor memory.

  The decoder 2 decodes the lossless compressed data 6 and generates linear PCM data. The linear PCM data generated by the decoder 2 has 16 bits for quantization bits and a frequency of 44.1 kHz.

  FIG. 2 is a block diagram showing the interpolation unit 3 in FIG. The interpolation unit 3 includes an upsampler 11, an input band division filter bank 12 as a division filter bank, an interpolation processing unit 13 as an interpolation processing unit, an output band synthesis filter bank 14 as a synthesis filter bank, and a variable equalizer. Each of the variable equalizer 15 for each band, the input signal frequency domain conversion unit 16, the prediction polynomial generation unit 17, and the power distribution determination unit 18 as an intensity distribution determination unit.

  The interpolation unit 3 is realized by a DSP (Digital Signal Processor) 7 as shown in FIG. The DSP 7 is a kind of computer that executes a program. For example, the interpolation unit 3 is realized by executing an interpolation program. Even if this interpolation program is provided to the user together with the portable audio player in a state stored in advance in a storage unit (not shown) of the DSP 7, it is provided to the user separately from the portable audio player and is not shown in the drawing of the DSP 7 by a predetermined storage process. May be stored in the storage unit. As a medium for providing the interpolation program to the user separately from the portable audio player, there are a transmission medium such as the Internet and a telephone communication network, and a computer-readable storage medium such as a CD and a semiconductor memory. The DSP 7 may realize the decoder 2 and the audio amplifier 4 together with the interpolation unit 3.

  Linear PCM data is input to the upsampler 11. The upsampler 11 upsamples the linear PCM data and generates linear PCM data having a sampling frequency of 88.2 kHz. The sampling frequency of linear PCM data is doubled.

  The input band division filter bank 12 generates information for each frequency band. Note that the input band division filter bank 12 includes a range of frequency components of the linear PCM data input to the upsampler 11 among the frequency components of the linear PCM data upsampled by the upsampler 11 (that is, the interpolation unit 3). Information for each frequency band is generated only within the input band (0 to 22 kHz).

  The input band division filter bank 12 generates information on a plurality of subbands from the generated information for each frequency band. A subband is each band when a certain band is divided into a plurality of bands having an equal bandwidth, for example. Here, the frequency component (0 to 22 kHz) of the linear PCM data upsampled by the upsampler 11 corresponds to a certain band. The subband information includes information on frequencies within the respective frequency bands.

  In addition, as a division filter bank that divides a frequency component of a certain band into a plurality of subbands of equal bandwidth, for example, various types such as QMF (Quadrature Mirror Filter), CQF (Conjugate Quadrature Filter), DTF, etc. There is a split filter bank.

  A plurality of subband information generated by the input band division filter bank 12 is input to the interpolation processing unit 13. The interpolation processing unit 13 sets a subband including the upper limit of the frequency component of the linear PCM data input to the upsampler 11 as a reference subband, and determines the correlation between this reference subband and all other subbands. Here, the upper limit of the frequency component of the linear PCM data input to the upsampler 11 is about 22 kHz. The interpolation processing unit 13 selects, for example, a 20 to 22 kHz subband as a reference subband.

  Further, the interpolation processing unit 13 interpolates the subband information determined to have a correlation into the input linear PCM data. The interpolation processing unit 13 determines that there is a correlation from the upper limit (here, about 22 kHz) of the frequency component of the linear PCM data input thereto to the Nyquist frequency (about 44 kHz here) of the linear PCM data input thereto. The subband information is interpolated repeatedly.

  The output band synthesis filter bank 14 generates linear PCM data from the information of the plurality of subbands interpolated by the interpolation processing unit 13 by synthesis. The output band synthesis filter bank 14 generates linear PCM data having frequency components up to the Nyquist frequency (here, about 44 kHz).

  The input signal frequency domain conversion unit 16 outputs data obtained by converting input data into the frequency domain. Linear PCM data input to the upsampler 11 is input to the input signal frequency domain conversion unit 16. The input signal frequency domain converter 16 converts linear PCM data into the frequency domain.

  The prediction polynomial generation unit 17 analyzes the frequency domain data converted by the input signal frequency domain conversion unit 16 and generates a prediction polynomial.

  The power distribution determination unit 18 receives the prediction polynomial generated by the prediction polynomial generation unit 17 and the plurality of subband information after interpolation generated by the interpolation processing unit 13. The power distribution determination unit 18 calculates an intensity correction coefficient for each subband so that the intensity distribution for each frequency band in the plurality of subband information after interpolation is a power intensity distribution according to the predicted polynomial.

  Data generated by the output band synthesis filter bank 14 is input to the band-specific variable equalizer 15. The band-specific variable equalizer 15 adjusts the intensity for each frequency band included in the input data based on the intensity correction coefficient for each subband calculated by the power distribution determination unit 18 for each subband. In this embodiment, the output band synthesis filter bank 14 generates data having a frequency component of 0 to 44 kHz as will be described later. The band-specific variable equalizer 15 may adjust, for example, the power intensity of 22 to 44 kHz in this data for each subband.

  Returning to FIG. An audio amplifier 4 is connected to the subsequent stage of the interpolation unit 3. Linear PCM data is input to the audio amplifier 4. The audio amplifier 4 generates an analog waveform signal from linear PCM data sampled at a frequency of 88.2 kHz. The amplitude of the analog waveform signal changes according to the quantization bit value of the linear PCM data. The audio amplifier 4 outputs the generated analog waveform signal to the headphone jack 5.

  A headphone plug (not shown) can be connected to the headphone jack 5. In FIG. 1, a headphone plug connected to a speaker 8 is attached to the headphone jack 5.

  Next, the operation of the portable audio player having the above configuration will be described.

  The decoder 2 reads the lossless compressed data 6 from the hard disk drive 1. The decoder 2 may read the lossless compressed data 6 of the music selected based on, for example, operation of an input key (not shown) of the portable audio player from the hard disk drive 1.

  The decoder 2 decodes the read lossless compressed data 6 to generate linear PCM data. The decoder 2 generates 16-bit linear PCM data sampled at a frequency of 44.1 kHz. The decoder 2 outputs the generated linear PCM data to the interpolation unit 3.

  The up-sampler 11 of the interpolation unit 3 up-samples the linear PCM data input from the decoder 2 and generates linear PCM data sampled at a sampling frequency of 88.2 kHz. For example, the upsampler 11 may output each quantized bit of 44.1 kHz input linear PCM data twice in succession under a sampling frequency of 88.2 kHz. As a result, linear PCM data sampled at a sampling frequency of 88.2 kHz is obtained.

  FIG. 3 is a schematic distribution diagram of frequency components of PCM data in the interpolation unit 3. FIG. 3A is a schematic distribution diagram of frequency components of linear PCM data sampled at a frequency of 44.1 kHz supplied from the decoder 2 to the interpolation unit 3. The linear PCM data output from the decoder 2 has a frequency component from 0 to about 22 kHz. FIG. 3B is a schematic distribution diagram of frequency components of linear PCM data sampled at a frequency of 88.2 kHz generated by the upsampler 11.

  As shown in FIGS. 3A and 3B, the linear PCM data includes an image component of the frequency component of the original linear PCM data by being upsampled to twice the frequency by the upsampler 11. . The image component is generated at about 22 to 44 kHz.

  The upsampler 11 supplies the generated linear PCM data to the input band division filter bank 12. First, the input band division filter bank 12 generates information for each predetermined frequency band from the input linear PCM data. The input band division filter bank 12, for example, decomposes the frequency components of the input linear PCM data into a plurality of bands, performs periodic addition for each band, and further executes DCT (Discrete Cosine Transform) processing. To do. Thereby, the input band division | segmentation filter bank 12 produces | generates the information for every frequency band about the band of 0-22 kHz.

  After generating information for each predetermined frequency band for the 0 to 22 kHz band, the input band division filter bank 12 generates information for a plurality of subbands. For example, the input band division filter bank 12 divides the band of 0 to 22 kHz into a plurality of bands having an equal bandwidth every 2 kHz and generates a plurality of subband information. In this case, 11 subbands are generated.

  FIG. 4 is a diagram schematically illustrating the division processing into subbands by the input band division filter bank 12. The frequency distribution in the upper part of FIG. 4 shows frequency components of linear PCM data supplied to the input band division filter bank 12. Linear PCM data has a frequency component of the original sound in a band of 0 to 22 kHz.

  The input band division filter bank 12 divides the band of the original sound of 0 to 22 kHz into 2 kHz bands, and generates a plurality of subbands for each 2 kHz band. In FIG. 4, the first subband corresponding to the 0 to 2 kHz band of the linear PCM data, the second subband corresponding to the 2 to 4 kHz band of the linear PCM data, and the 18 to 20 kHz band of the linear PCM data. A corresponding tenth subband and an eleventh subband corresponding to a 20 to 22 kHz band of linear PCM data are illustrated.

  The input band division filter bank 12 supplies the generated information of the plurality of subbands to the interpolation processing unit 13. First, the interpolation processing unit 13 selects a subband on the highest frequency side among the supplied subbands as a reference subband. In the case of a plurality of subbands illustrated in FIG. 4, the interpolation processing unit 13 selects the eleventh subband corresponding to the frequency component of 20 to 22 kHz of the original sound as the reference subband.

  After selecting the reference subband, the interpolation processing unit 13 compares the reference subband with all the other subbands (here, the first subband to the tenth subband) in order, and determines the respective correlations. . The frequency component of the tenth subband shown in FIG. 4 includes three overtone components in an obvious manner, and is similar to the frequency component of the eleventh subband. The frequency components of the first subband and the second subband are not as similar to the eleventh subband as the tenth subband. In this case, for example, the interpolation processing unit 13 determines that the tenth subband is correlated, and determines that the first subband and the second subband are not correlated.

  Specifically, for example, the interpolation processing unit 13 may perform the following processing and determine the correlation between the reference subband and all other subbands based on the similarity of the frequency components. That is, the interpolation processing unit 13 first inputs the characteristic values of the reference subband and another subband to a predetermined correlation function. Accordingly, the interpolation processing unit 13 obtains 0 as the correlation value when there is no correlation between the subbands, and obtains 1 as the correlation value when there is 100% correlation between the subbands. In this case, a correlation value within a range of 0 to 1 is obtained according to the degree of correlation between subbands. Next, the interpolation processing unit 13 compares the correlation values of the two subbands obtained by a predetermined correlation function with a predetermined threshold value such as 0.8. Then, the interpolation processing unit 13 determines that there is a correlation when the correlation value is greater than or equal to a predetermined threshold, and determines that there is no correlation when the correlation value is smaller than the predetermined threshold.

  After determining the correlation between the subbands, the interpolation processing unit 13 starts interpolation processing. The interpolation processing unit 13 sets the whole of the plurality of subbands determined to have a correlation as a band having one correlation, and sets the entire band having the correlation from the upper limit of the frequency component of the linear PCM data to the linear PCM data. Interpolate up to the Nyquist frequency. The interpolation processing unit 13 repeatedly interpolates information on a correlated frequency band from the upper limit of the frequency component of the linear PCM data to the Nyquist frequency of the linear PCM data.

  FIG. 3C is a schematic distribution diagram of frequency components of a plurality of subband information generated by the interpolation processing unit 13. The frequency component of FIG. 3C repeats the frequency band of the original sound, which is determined to be correlated with the frequency component of FIG. Interpolated. Accordingly, the plurality of subband information generated by the interpolation processing unit 13 includes information on frequency components from 0 to 44 kHz.

  The interpolation processing unit 13 supplies a plurality of subband information after interpolation to the output band synthesis filter bank 14. The output band synthesis filter bank 14 generates linear PCM data from a plurality of supplied subband information by a synthesis process of a plurality of subbands. The output band synthesis filter bank 14 generates linear PCM data having a sampling frequency of 88.2 kHz and having frequency components from 0 to 44 kHz.

  As shown in FIG. 2, the linear PCM data output from the decoder 2 to the interpolation unit 3 is supplied to the input signal frequency domain conversion unit 16 in the upper stage of FIG. The input signal frequency domain converter 16 converts the input linear PCM data into the frequency domain.

  The prediction polynomial generation unit 17 analyzes the frequency domain data converted by the input signal frequency domain conversion unit 16 and generates a prediction polynomial. Specifically, for example, the prediction polynomial generation unit 17 first analyzes the frequency domain data converted by the input signal frequency domain conversion unit 16, and the power of the linear PCM data input to the input signal frequency domain conversion unit 16. Get strength. Next, the power distribution determination unit 18 predicts the frequency characteristics of the analyzed linear PCM data based on the power intensity and a predetermined auditory model set in advance, and generates a prediction polynomial for the frequency characteristics.

  Note that frequency components such as music generally have frequency characteristics in which the power intensity is high on the low frequency side and the power intensity decreases as the frequency becomes higher. In addition, the power intensity generally decreases gradually as the frequency increases. Therefore, the power distribution determination unit 18 predicts a polynomial having such frequency characteristics.

  The power distribution determination unit 18 receives the prediction polynomial generated by the prediction polynomial generation unit 17 and the plurality of subband information after interpolation generated by the interpolation processing unit 13. The power distribution determination unit 18 calculates an intensity correction coefficient for each frequency band so that the intensity distribution for each frequency band in the plurality of subband information after interpolation becomes a power intensity distribution according to the predicted polynomial.

  Specifically, for example, the power distribution determination unit 18 first, based on the intensity distribution of the band of 0 to 22 kHz in the intensity distribution for each frequency band in the plurality of subband information after interpolation and the prediction polynomial, The intensity correction coefficient for each subband is calculated so that the power intensity for each subband of 22 to 44 kHz, which is the interpolated band, follows the calculated power intensity distribution.

  The intensity correction coefficient for each subband calculated by the power distribution determination unit 18 is supplied to the band-specific variable equalizer 15. In addition to this, linear PCM data based on FIG. 3C generated by the output band synthesis filter bank 14 is input to the variable equalizer 15 for each band.

  The band-specific variable equalizer 15 multiplies the linear PCM data supplied from the output band synthesis filter bank 14 by the intensity correction coefficient of each subband for each frequency band corresponding to the subband.

  FIG. 3D is a schematic distribution diagram of frequency components of linear PCM data generated by the band-specific variable equalizer 15. In the linear PCM data generated by the band-specific variable equalizer 15, the power intensity of the frequency component of about 22 to 44 kHz is reduced. Further, the power intensity of the frequency component lower than about 22 kHz that is the original sound and the power intensity of the frequency component higher than about 22 kHz that is the interpolated sound are gently continuous.

  Further, as apparent from comparing the schematic distribution chart of FIG. 3D with the schematic distribution chart generated by the output band synthesis filter bank 14 of FIG. The frequency component has a lower intensity distribution with respect to the frequency component of the audio waveform data input to the upsampler 11 as a reference.

  The band-specific variable equalizer 15 outputs the generated linear PCM data to the audio amplifier 4. The linear PCM data output from the band-specific variable equalizer 15 of the audible component interpolation unit 3 is linear PCM data sampled at a frequency of 88.2 kHz. Moreover, it has a frequency component of 0 to 44 kHz.

  The audio amplifier 4 generates an analog waveform signal from linear PCM data having a sampling frequency of 88.2 kHz. The amplitude of the analog waveform signal corresponds to the data of linear PCM data. The audio amplifier 4 outputs the generated analog waveform signal to the speaker 8 via the headphone jack 5. The speaker 8 vibrates a diaphragm (not shown) according to the waveform of the supplied analog waveform signal. Thus, the speaker 8 outputs a sound having a frequency component of 0 to 44 kHz based on the linear PCM data having a sampling frequency of 88.2 kHz.

  FIG. 5 is a diagram illustrating an example of the spectral intensity distribution of the linear PCM data output from the decoder 2 and the spectral intensity distribution of the linear PCM data output from the interpolation unit 3 to the audio amplifier 4. The spectral intensity distribution shown in the upper part of FIG. 5 is a spectral intensity distribution of linear PCM data output from the decoder 2 to the interpolation unit 3. The spectral intensity distribution shown in the lower part of FIG. 5 is a spectral intensity distribution of linear PCM data output from the interpolation unit 3 to the audio amplifier 4.

  As shown in FIG. 5, the linear PCM data output from the decoder 2 to the interpolation unit 3 has only frequency components in the audible band of 0 to 22 kHz. On the other hand, the linear PCM data output from the interpolation unit 3 to the audio amplifier 4 has a frequency component of 0 to 44 kHz. A frequency component outside the audible band of about 22 to 44 kHz is interpolated.

  Further, the spectral intensity distribution of the linear PCM data output from the interpolation unit 3 to the audio amplifier 4 naturally decreases as the frequency increases as a whole, and gradually decreases from the low frequency side to the high frequency side. There is a harmony between the intensity distribution of the frequency components of the original sound and the intensity distribution of the interpolated frequency components.

  Further, in the linear PCM data output from the interpolation unit 3 to the audio amplifier 4, the interpolated frequency component of about 22 to 44 kHz includes a harmonic component similar to about 12 to 22 kHz. There is a natural harmony between the intensity distribution of the harmonic component of the original sound and the intensity distribution of the interpolated harmonic component.

  In addition, when audio critics evaluated the sound quality that was appropriately interpolated with such overtone components, the sound quality improved, unlike when high frequency components with noisy sound quality were interpolated outside the audible band. I was able to conclude.

  As described above, in the first embodiment, the interpolation processing unit 13 subbands (tenth in FIG. 4) of the band corresponding to the highest frequency side of the 44.1 kHz linear PCM data input to the upsampler 11. The correlation between subbands is determined with reference to one subband. Then, the interpolation processing unit 13 obtains information for each frequency of the correlated subband (for example, the subband on the high frequency side of the audible band such as 12 to 22 kHz) from the high frequency band (for example, 22 to 44 kHz) outside the audible band. Interpolate to In general, a frequency band that is 10 kHz lower than the upper limit of the audible band includes notably overtone components such as musical instruments. By interpolating the frequency component containing the harmonic component remarkably in this way outside the audible band, the same harmonic component as that on the high frequency side of the audible band is interpolated in the high frequency band outside the audible band.

  Further, the interpolation processing unit 13 performs linear PCM up-sampled by the up-sampler 11 from the upper limit (for example, 22 kHz) of the frequency component of the linear PCM data of 44.1 kHz input to the up-sampler 11 for the correlated subbands. Interpolation is repeated until the Nyquist frequency of the data (for example, 44 kHz). Overtone components are interpolated in the upsampled linear PCM data up to the Nyquist frequency. Overtone components are continuously interpolated from the audible region to the Nyquist frequency in a natural harmonious state.

  As a result, in the linear PCM data generated by the decoder 2, a harmonic component that is continuously harmonized with the high frequency side of the audible band is interpolated in the high frequency band outside the audible band. The audio amplifier 4 can generate an audio waveform signal with good sound quality. Sound quality is improved.

  In the first embodiment, the intensity of the frequency component outside the audible band interpolated by the interpolation processing unit 13 is based on the frequency component of the linear PCM data input to the upsampler 11 by the variable equalizer 15 for each band. The intensity distribution is lower than that. The frequency distribution after the interpolation is a natural intensity distribution that decreases from the low frequency side to the high frequency side. The distribution of overtone components that are prominently included from the high frequency portion of the audible region to the outside of the audible region also naturally decreases from the low frequency side to the high frequency side. In this way, harmonic harmony components are harmonized in the entire band of the audio waveform data, so that better sound quality can be improved.

  Further, in the first embodiment, the input band division filter bank 12 performs subband information (information for each frequency band) only within the frequency component range (0 to 22 kHz) of the linear PCM data input to the upsampler 11. ) Is generated. Linear PCM data generated by the upsampler 11 is input to the input band division filter bank 12. The linear PCM data generated by the upsampler 11 includes the image component in addition to the frequency component of the original sound. The image component is included on the higher frequency side than the frequency component of the original sound. By making the processing of the input band division filter bank 12 only the above-described original sound band, it is necessary to provide a low-pass filter between the upsampler 11 and the input band division filter bank 12 as in the second embodiment to be described later. Disappear. In addition, the input band division filter bank 12 has only a frequency component on the low frequency side (that is, 0 to 22 kHz) of the frequency components (0 to 44 kHz in this case) of the linear PCM data generated by the upsampler 11. What is necessary is just to process, and the processing load is reduced.

  In the first embodiment, the input band division filter bank 12 divides only the frequency component range (0 to 22 kHz) of the linear PCM data input to the upsampler 11 into a plurality of subbands. . In addition to this, for example, the division filter bank that divides linear PCM data into a plurality of subbands, all frequency components below the Nyquist frequency (for example, 44.2 kHz) of the linear PCM data generated by the upsampler 11 are You may divide | segment into a subband.

  FIG. 6 is a schematic distribution diagram of frequency components of PCM data in the interpolation unit 3 in this modification. FIG. 6A is a diagram illustrating frequency components of linear PCM data input to the upsampler 11. FIG. 6B is a diagram illustrating frequency components of linear PCM data generated by the upsampler 11. FIG. 6C is a diagram illustrating frequency components of linear PCM data generated by the interpolation processing unit 13. FIG. 6D is a diagram illustrating frequency components of linear PCM data generated by the band-specific variable equalizer 15.

  In this modification, as shown in FIG. 6B, the division filter bank generates subband information up to the Nyquist frequency (44 kHz), and the interpolation processing unit 13 determines the correlation for all the subbands. . At this time, the interpolation processing unit 13 may determine the correlation using a subband corresponding to the highest frequency band in the original sound band (for example, a subband of 20 to 22 kHz) as a reference subband. Then, as shown in FIG. 6C, the interpolation processing unit 13 interpolates the subband on the highest frequency side and the subband correlated therewith outside the audible band.

  As shown in FIG. 6B, the linear PCM data generated by the upsampler 11 includes an image component obtained by mapping the original sound component. This image component also includes a harmonic component. The interpolation processing unit 13 determines that the subband of the image component is a correlated subband. Therefore, the interpolation processing unit 13 determines that the subband including the image of the harmonic component is a correlated subband, and interpolates it outside the audible band. The number of subbands for which the interpolation processing unit 13 determines that there is a correlation is approximately doubled. As shown in FIG. 6C, when the Nyquist frequency is filled with correlated subbands, the number of repetitions can be reduced. In FIG. 3C, the number of repetitions is about 2.4 times, but in FIG. 6C, it is about 1.2 times. It is halved.

Embodiment 2. FIG.
FIG. 7 is a block diagram showing the interpolation unit 3 of the portable audio player according to Embodiment 2 of the present invention. The interpolation unit 3 includes an LPF (low-pass filter) 21 between the upsampler 11 and the input band division filter bank 12.

  The LPF 21 extracts a low frequency component from the linear PCM data input from the upsampler 11 and supplies the low frequency component to the input band division filter bank 12. A frequency component equal to or lower than the Nyquist frequency (about 22 kHz) of linear PCM data having a sampling frequency of 44.1 kHz input to the upsampler 11 is extracted.

  The components shown in FIG. 7 other than those described above are the same as the components having the same names in FIG. 2 according to the first embodiment, and the description thereof will be omitted by assigning the same reference numerals. The configuration other than the interpolation unit 3 of the portable audio player is the same as that in FIG. 1 of the first embodiment, and the same names and reference numerals as those in the first embodiment are given and description thereof is omitted.

  Next, the operation of the portable audio player having the above configuration will be described.

  The upsampler 11 up-samples the linear PCM data input from the decoder 2 and generates linear PCM data sampled at a sampling frequency of 88.2 kHz.

  Linear PCM data generated by upsampling by the upsampler 11 is supplied to the LPF 21. The LPF 21 extracts a frequency component of about 22 kHz or less and supplies it to the input band division filter bank 12. The input band division filter bank 12 first generates information for each frequency band for the band of 0 to 22 kHz from the input linear PCM data, and then generates a plurality of subband information.

  The operations of the portable audio player according to the second embodiment other than those described above are the same as those of the first embodiment, and a description thereof will be omitted.

  As described above, in the second embodiment, the LPF 21 provided between the upsampler 11 and the input band division filter bank 12 extracts a frequency component of about 22 kHz or less. Therefore, only the solid line portion in the frequency component of FIG. 3B can be supplied to the input band division filter bank 12.

  In the second embodiment, an input band division filter bank 12 that generates a plurality of subband information for the band of 0 to 22 kHz is provided in the subsequent stage of the LPF 21. In addition, for example, a division filter bank that generates a plurality of pieces of subband information for the band of 0 to 44 kHz may be provided in the subsequent stage of the LPF 21. In the case of this modification, the band divided by the division filter bank into a plurality of subbands and the band synthesized by the output band synthesis filter bank 14 can be matched. As a combination of the division filter bank and the synthesis filter bank, a general combination having the same band can be used.

  Each of the above embodiments is an example of a preferred embodiment of the present invention, but the present invention is not limited to this, and various modifications and changes can be made without departing from the scope of the invention. It is.

  For example, in each of the above embodiments, the input band division filter bank 12 divides the frequency components of the linear PCM data input thereto into a plurality of subbands for each fixed bandwidth. The bandwidths of the plurality of subbands are the same. In addition to this, for example, the input band division filter bank 12 may be divided into a plurality of subbands having different bandwidths for each frequency band. As a method of making the subband bandwidth different for each frequency band, for example, there is a two-division method. That is, in this two-division method, for example, the frequency component of the input linear PCM data is first divided in half, and then the frequency component on the high-frequency side in the half divided is further divided in half. The frequency components on the high frequency side in the two divided bands are sequentially divided into two. Thereby, the bandwidth of the subband becomes narrower toward the high frequency side. The output band synthesis filter bank 14 has a configuration corresponding to the division of the input band division filter bank 12.

  In each of the above embodiments, the interpolation processing unit 13 determines the correlation for all of the plurality of subbands. In addition, for example, the interpolation processing unit 13 may determine the correlation only for the subbands within a certain band. In CD linear PCM data, overtone components of acoustic instruments tend to appear remarkably in frequency components of about 10 kHz or higher. Therefore, the interpolation processing unit 13 may determine the correlation only for subbands of about 10 kHz or higher. In addition, in music such as pops, a harmonic component may be manifested in a band of 16 kHz or higher. Therefore, the interpolation processing unit 13 may determine the correlation only for subbands of about 16 kHz or higher.

  Furthermore, for example, the interpolation processing unit 13 may change the frequency range for determining the correlation. Then, for example, based on an instruction from a control unit (not shown), the interpolation processing unit 13 may determine the correlation with respect to subbands in a frequency band selected from, for example, 10 kHz, 12 kHz, 14 kHz, and 16 kHz. .

  Further, the control unit may specify the frequency band based on tag data such as the genre of music included in the lossless compressed data 6 of the music, the genre of the acoustic effect of the equalizer set by the user, or the like. Good. For example, if the control unit determines that the genre of the music to be played is classic based on the tag data or the equalizer sound effect setting, it specifies 10 kHz or more, and if it determines that it is pop, it specifies 16 kHz. Good.

  In each of the above embodiments, the hard disk drive 1 stores the lossless compressed data 6 of the music, and the decoder 2 generates linear PCM data from the lossless compressed data 6. In addition, for example, the hard disk drive 1 may store linear PCM data for CD, and the decoder 2 may read the linear PCM data. Furthermore, for example, the hard disk drive 1 is compressed by an irreversible compression method such as MP3 (MPEG Audio Layer-3), AAC (Advanced Audio Coding), WMA (Windows (registered trademark) Media Audio) or the like. Data may be stored, and the decoder 2 may generate linear PCM data from this compressed data.

  In compressed data such as MP3, the amount of data is reduced while suppressing deterioration in sound quality by thinning out data in the high sound range (approximately 16 kHz or more) of linear PCM data. Therefore, if the decoder 2 simply decodes the compressed data such as MP3, the linear PCM data generated by the decoder 2 does not include a frequency component of 16 kHz or higher. Therefore, it is better for the decoder 2 that decodes compressed data such as MP3 to decode the compressed data and interpolate frequency components up to 22 kHz. For this interpolation processing, it is possible to use a block configuration substantially similar to the audible out-of-band interpolation unit 3 shown in FIG.

  In each of the above embodiments, linear PCM data having a frequency component up to the Nyquist frequency (for example, about 22 kHz) is input to the interpolation unit 3, and this linear PCM data is divided into a plurality of subbands. Interpolates frequency components above the Nyquist frequency. In addition to this, for example, linear PCM data whose upper limit of the frequency component is lower than the Nyquist frequency, such as linear PCM data based on MP3, is input to the interpolation unit 3, and this linear PCM data is divided into a plurality of subbands. The high frequency component may be interpolated. Thus, when linear PCM data whose upper limit of the frequency component is lower than the Nyquist frequency is input, the bandwidth of the band that the input band dividing filter bank 12 of the interpolation unit 3 divides into subbands is narrowed. Further, the bandwidth of the band to be interpolated by the output band synthesis filter bank of the interpolation unit 3 is widened.

  In addition, when interpolating high frequency components outside the audible band in the compressed data such as MP3 in this way, by first interpolating the frequency component outside the audible band and then interpolating the frequency component outside the audible band, As a configuration after the decoder 2 of the portable audio player, a configuration for reproducing the CD and the lossless compressed data 6 can be commonly used. It is not necessary to design the structure after the decoder 2 of the portable audio player for each sound source. By designing only the structure before the decoder 2, the portable audio player can reproduce MP3 and the like with high sound quality. The types of data that the portable audio player can reproduce with high sound quality can be easily increased.

  In each of the above embodiments, the interpolation unit 3 shown in FIG. 2 and the like interpolates frequency components in a band of 22 kHz or higher. In addition to this, for example, the interpolation unit 3 shown in FIG. 2 or the like may interpolate a frequency component of 22 kHz or less, for example, an audible band. For example, when linear PCM data based on MP3 whose upper limit of frequency components is about 16 kHz is input to the interpolation unit 3 shown in FIG. 2 and the like, the interpolation unit 3 interpolates frequency components of 16 to 44 kHz. May be. In addition, the interpolating unit 3 shown in FIG. 2 or the like may be input with sampled audio data transmitted with a limited band such as FM radio or AM radio other than MP3.

  In each of the above embodiments, the interpolation unit 3 interpolates high frequency components outside the audible band, but the interpolation device of the present invention may interpolate high frequency components in the audible band.

  In the above embodiments, the audio playback device is a portable audio player. In addition to this, for example, the audio reproducing apparatus includes a car audio system, a car navigation system, a home audio system, a reproducing apparatus such as a CD and a DVD, a portable information terminal such as a mobile phone terminal and a PDA, and a personal computer having an audio output function. It may be.

  The present invention can be used for portable hard disk players and the like.

FIG. 1 is a block diagram showing a portable audio player according to Embodiment 1 of the present invention. FIG. 2 is a block diagram showing the interpolation unit in FIG. FIG. 3 is a schematic distribution diagram of frequency components of PCM data in the interpolation unit. FIG. 4 is a diagram schematically illustrating subband division processing by the input band division filter bank in FIG. FIG. 5 is a diagram illustrating an example of a spectral intensity distribution of linear PCM data output from the decoder and a spectral intensity distribution of linear PCM data output from the interpolation unit to the audio amplifier. FIG. 6 is a schematic distribution diagram of frequency components of PCM data in the interpolation unit 3 in a modification of the first embodiment. FIG. 7 is a block diagram showing an interpolation unit of the portable audio player according to Embodiment 2 of the present invention.

Explanation of symbols

2 Decoder 3 Interpolator (interpolator)
4 Audio amplifier 11 Upsampler 12 Input band division filter bank (division filter bank)
13 Interpolation processing unit (interpolation processing means)
14 Output Band Synthesis Filter Bank (Synthesis Filter Bank)
15 Variable equalizer by band (variable equalizer)
18 Power distribution determination unit (intensity distribution determination means)
21 LPF (low pass filter)

Claims (10)

An upsampler that receives input audio waveform data having an audible frequency component and raises the sampling frequency of the audio waveform data;
A divided filter bank that generates a plurality of subbands for each predetermined frequency band within a range of frequency components of at least the input audio waveform data from the audio waveform data upsampled by the upsampler;
The correlation between the plurality of subbands for each predetermined frequency band generated by the division filter bank and the highest frequency subband within the range of the frequency component of the input audio waveform data is determined, and the highest frequency subband is determined. Based on the frequency components of the predetermined frequency band of the band and the subbands correlated therewith, the entire plurality of subbands determined to be correlated are defined as one correlated band, and the entire correlated band is defined as Interpolation processing means for interpolating as a frequency component above the upper limit of the frequency component of the input audio waveform data ;
A synthesis filter bank that generates audio waveform data from the frequency components of the predetermined frequency band of the audio waveform data interpolated with the frequency components of the predetermined frequency band of the correlated subbands;
An interpolation apparatus characterized by comprising: The audio waveform data input to the upsampler is such that the upper limit of the frequency component is at least 10 kHz lower than the upper limit of the audible band,
The interpolating apparatus according to claim 1. The division filter bank generates a plurality of subbands for each predetermined frequency band only within the frequency component range of the input audio waveform data among the frequency components of the audio waveform data upsampled by the upsampler. The interpolating apparatus according to claim 1 or 2, characterized in that: A low-pass filter that extracts a frequency component below the upper limit of the frequency component of the input audio waveform data from the audio waveform data up-sampled by the up-sampler;
The interpolation apparatus according to claim 1, wherein the division filter bank generates a plurality of subbands for each predetermined frequency band for the frequency component extracted by the low-pass filter. The divided filter bank generates a plurality of subbands for each predetermined frequency band for all frequency components below the Nyquist frequency in the frequency components of the audio waveform data upsampled by the upsampler.
The interpolation apparatus according to claim 1 or 2. The interpolation processing means repeatedly interpolates the correlated subbands to fill from the upper limit of the frequency component of the input audio waveform data to the Nyquist frequency of the audio waveform data upsampled by the upsampler,
Interpolator according to any one among claims 1 to 5, wherein. Intensity distribution determination means for predicting an intensity lower than the intensity of the input audio waveform data input to the upsampler as the intensity of the component to be interpolated;
A variable equalizer that changes the intensity of at least the interpolated frequency component among the frequency components of the audio waveform data generated by the synthesis filter bank based on the predicted intensity;
Interpolator according to any one among claims 1 to 6, characterized in that it comprises a. A decoder for generating audio waveform data of a predetermined sampling frequency having a frequency component of an audible band;
  The interpolator according to any one of claims 1 to 7, which interpolates a frequency component on a high frequency side using the audio waveform data generated by the decoder as input audio waveform data,
  An audio amplifier that generates an audio waveform signal from the audio waveform data interpolated by the interpolation device;
  An audio playback apparatus comprising: Increasing the sampling frequency of audio waveform data having an audible frequency component;
  Generating a plurality of subbands for each predetermined frequency band, at least within a range of frequency components of the input audio waveform data, from the audio waveform data having the increased sampling frequency;
  The correlation between the plurality of subbands for each predetermined frequency band generated by the division filter bank and the highest frequency subband within the range of the frequency component of the input audio waveform data is determined, and the highest frequency subband is determined. Based on the frequency components of the predetermined frequency band of the band and the subbands correlated therewith, the entire plurality of subbands determined to be correlated are defined as one correlated band, and the entire correlated band is defined as Interpolating as a frequency component above the upper limit of the frequency component of the input audio waveform data;
  Generating audio waveform data from the frequency components of the predetermined frequency band of the audio waveform data interpolated with the frequency components of the predetermined frequency band of the correlated subbands;
  An interpolation method characterized by comprising:   Computer
  An upsampler that receives input audio waveform data having an audible frequency component and raises the sampling frequency of the audio waveform data;
  A divided filter bank that generates a plurality of subbands for each predetermined frequency band within a range of frequency components of at least the input audio waveform data from the audio waveform data upsampled by the upsampler;
  The correlation between the plurality of subbands for each predetermined frequency band generated by the division filter bank and the highest frequency subband within the range of the frequency component of the input audio waveform data is determined, and the highest frequency subband is determined. Based on the frequency components of the predetermined frequency band of the band and the subbands correlated therewith, the entire plurality of subbands determined to be correlated are defined as one correlated band, and the entire correlated band is defined as Interpolation processing means for interpolating as a frequency component above the upper limit of the frequency component of the input audio waveform data;
  A synthesis filter bank that generates audio waveform data from the frequency components of the predetermined frequency band of the audio waveform data interpolated with the frequency components of the predetermined frequency band of the correlated subbands;
  Interpolation program characterized by functioning.

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