JP2006064884A - Audio equipment - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To reasonably reproduce audio signals from various sources and to complement and reproduce a signal of a frequency band lost in a stage of encoding. <P>SOLUTION: A client control IC 12 decodes encoded audio data and acquires text data of the file name etc., of the audio data. An audio DSP 14 processes the decoded audio data to reproduce an audio signal. A system microcomputer 11 receives the text data and outputs them for display, and also controls a listening mode of the audio DSP 14 according to an input indication. The client control IC 12 acquires the extension of the file name and supplies it to the audio DSP 14 through the system microcomputer 11. The audio DSP 14 specifies an encoding system from the extension and performs sound field complement processing for complementing the signal of the frequency band lost by the encoding system. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to an audio apparatus.

  In recent years, it has become popular to distribute audio data representing audio such as music via a network such as the Internet or to record and use it on a recording medium such as an MD (Mini Disk). Audio data distributed over a network or recorded on a recording medium is generally limited to a certain amount of music to be supplied in order to avoid an increase in the amount of data and an increase in occupied bandwidth due to an excessively wide band. Components above the frequency have been removed.

  For example, in audio data in MP3 (MPEG1 audio layer 3) format, frequency components of about 16 kilohertz or more are removed. Further, in the audio data in the ATRAC3 (Adaptive Transform Acoustic Coding 3) format, frequency components of about 14 kilohertz or more are removed. In addition, WMA, OGG Vorbis, and the like are generally known. However, even with these compression methods, a missing frequency band may occur depending on the mode and the bit rate.

  As described above, music or the like from which a frequency component equal to or higher than a certain value is removed usually has a degraded sound quality as compared to original music or the like. Therefore, it is conceivable to add a signal in place of the removed frequency component. As a technique for this purpose, there is a technique disclosed in Patent Document 1.

The technique disclosed in Patent Document 1 multiplies an output audio signal obtained by passing a PCM (Pulse Code Modulation) digital audio signal through a low-pass filter by a signal including an absolute value component of the output signal. It is a technique of generating.
JP-A-7-93900

  However, the sound signal representing music is generally about several hundred hertz (100 Hz or more and less than 1 kHz) while a component of 10 kHz or more is attenuated as the frequency increases due to harmonics of an electronic musical instrument or a human voice (vocal). ) Has a spectral distribution that includes many peaks arranged at intervals. In the case of a voice signal representing voice transmitted through a telephone line, generally, a component of 4 kHz or higher shows a spectrum distribution having the same characteristics.

  For this reason, in the audio signal reproduction device of Patent Document 1 that only generates harmonics by distorting the waveform of the low frequency component of the output audio signal using an absolute value circuit or the like, a signal having a spectrum close to the original sound is generated. Can't get.

  In addition, there is a problem that such a method of adding a signal in place of the removed frequency component to a signal to be reproduced cannot be applied to an audio system that can use various sources.

This point will be described with reference to FIG.
FIG. 7 shows the configuration of a conventional audio system. In FIG. 7, the content storage unit 33 stores audio data encoded and compressed according to various standards, and provides arbitrary audio data to the decoder 32. The decoder 32 decodes the encoded audio data and supplies it to the amplifier unit 31.

  In such a configuration, it is functionally possible to add a high-frequency component by the decoder 32, but the decoding process and the process of adding the high-frequency component all become a load on the decoder 32 and require a large processing capacity. It becomes.

  On the other hand, it is logically possible to add a high frequency component by the amplifier unit 31. However, since the decoded PCM data is supplied to the amplifier unit 31, which method the original signal is compressed in, or in which frequency range the missing frequency component is, Therefore, proper function control cannot be performed. In particular, the frequency band where the audio signal is missing differs depending on the compression format, and even if the audio signal is compression-encoded in the same compression format, the frequency band that is missing depends on the bit rate at the time of compression. Will change. Therefore, if interpolation processing is simply performed on a speech signal that has been compression-encoded, sound quality may be deteriorated. That is, when an interpolation signal is to be added to the interpolated band, depending on the compression format or bit rate, there may be an audio signal in the frequency band that is the interpolated band. Otherwise, the sound quality will deteriorate. For example, when compressing and encoding an audio signal in MP3 format, the upper limit of the frequency band where the audio signal exists is about 16 kHz when encoded at 128 kbps, but the audio signal exists up to about 22 kHz when encoded at 256 kbps. To do. Therefore, if the above-described interpolation signal is added to the MP3 format audio signal encoded at 256 kHz, the sound quality deteriorates.

The present invention has been made in view of the above circumstances, and an audio device capable of reproducing an audio signal in which a signal in a part of a frequency band is lost in a process of encoding or compression in a natural sound field. The purpose is to provide.
It is another object of the present invention to provide an audio apparatus that can easily reproduce audio signals from various sources and supplement and reproduce signals in a frequency band that has been lost during compression. .

In order to achieve the above object, an audio device according to the first aspect of the present invention provides:
Decoding means for decoding and outputting the encoded audio data and obtaining information relating to the audio data, reproduction means for processing the decoded audio data and reproducing an audio signal, and audio data from the decoding means In an audio system comprising: control means for receiving information and outputting it for display and controlling the operation of the reproduction means based on an input instruction;
The decoding means supplies information corresponding to the encoding method of the audio data to the control means,
The control means supplies information corresponding to the information corresponding to the encoding method from the decoding means to the reproducing means,
The reproduction unit includes a complementing unit that executes a sound field complementing process that complements a signal in a frequency band that is missing in the encoding scheme, based on information from the control unit.
It is characterized by that.

Any of the decoding means, the control means, and the reproduction means includes means for obtaining a control parameter for sound field interpolation processing corresponding to the encoding method,
The reproducing means includes a complementing means for executing a sound field complementing process for complementing a signal in a frequency band corresponding to the encoding method according to a control parameter.

The signal transmission format itself is arbitrary. For example, the decoding unit may obtain a control parameter corresponding to the encoding method, and may transmit this to the reproducing apparatus via the control unit. Alternatively, the decoding unit may transmit information specifying the encoding method to the control unit, and the control unit may obtain a control parameter and notify the reproduction apparatus of the control parameter.
Also, the decoding means transmits information (for example, file format, extension, and / or bit rate) specifying the encoding method to the control unit, and the control unit notifies the playback apparatus as it is or in another signal form. Also good.
Further, the decoding means transmits information specifying the encoding method (for example, file format, extension and / or bit rate) to the control unit, and the control unit converts it to information specifying the missing frequency and reproduces it. May change it into an operating parameter.

The interpolation means includes
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal that represents a speech that is subject to spectrum interpolation;
A filter that extracts a component in a second band on a higher frequency side than the first band occupied by the interpolated signal in the interpolation signal;
An adder for generating an output signal representing the sum of the interpolated signal and the component extracted by the filter;
It is preferable to provide.

  The information corresponding to the encoding method preferably includes a file format and / or a bit rate.

  According to the above configuration, the decoding unit can execute the decoding process and the reproducing unit can execute the sound field addition process, and the processing load can be distributed. In addition, information necessary for sound field complementation can be transmitted through an originally necessary information transmission route such as decoding means and control means, and control means and reproduction means.

  Hereinafter, an AV apparatus including the audio apparatus according to the embodiment of the present invention will be described.

  As shown in FIG. 1, a content reproduction system according to an embodiment of the present invention includes a system microcomputer 11, a client control IC (Integrated Circuit) 12, a DIR (Digital Interface Receiver) 13, and an audio DSP (Digital Signal Processor). ) 14, amplifier unit 15, speaker unit 16, key control unit 17, PC (Personal Computer) 18, router 19, memory card 20, slot 21, LAN (Local Area Network) 22, The LAN controller 23, a display unit 24, a display control microcomputer 25, and a LAN cable connection terminal 26 are included.

  Of the above configuration, the system microcomputer 11, the client control IC 12, the DIR 13, the audio DSP 14, the amplifier unit 15, the key control unit 17, the slot 21, the LAN controller 23, the display unit 24, and the display control microcomputer. 25 and the LAN cable connection terminal 26 are arranged in the AV apparatus 1.

  The system microcomputer 11 controls the entire operation of the AV system and the entire AV device 1, and includes a CPU (Central Processing Unit), a ROM (Read Only Memory), a RAM (Random Access Memory), and the like.

  The client control IC 12 determines the file name of the digital audio data supplied from the PC 18 or the memory card 20, for example, determines the encoding method of the digital data by determining the extension and the bit rate, This is decoded into PCM data by a decoding method corresponding to the determined encoding method and supplied to the DIR 13.

  As shown in FIG. 2, the client control IC 12 selects one of the PC 18 and the memory card 20 via the LAN as a content source and performs a communication process with the selected source. A decoding unit 52 that decodes supplied encoded data into a PCM signal and a system control unit 53 are provided.

  Further, the system control unit 53 determines the encoding method of the audio data supplied from the source via the selector unit 51 from the extension of the file name, and designates the decoding method to the decoding unit 52.

  The system control unit 53 communicates with the system microcomputer 11 and switches the selector unit 51 in accordance with an instruction from the system microcomputer 11. Further, the system control unit 53 transmits attribute information (file format, sampling frequency, etc.) and text information (file name, performance time, performer, etc.) accompanying the audio data to the system microcomputer 11.

  The DIR 13 supplies the PCM data supplied from the client control IC 12 to the audio DSP 14.

  As shown in FIG. 2, the audio DSP 14 functionally includes a high frequency signal adding unit 61 and a listening mode processing unit 62.

When the music data stored in the PC 18 or the memory card 20 is in a format in which the high band frequency portion is removed, the high band signal adding unit 61 adds the high band frequency component to the original signal and outputs it.
When the music data stored in the PC 18 or the memory card 20 is in a format from which the high-band frequency portion is removed, the high-frequency signal adding unit 61 adds a wide-band frequency component to the original signal and outputs it. To do.
The high-frequency signal adding unit 61 is functionally configured to include a delay unit 71, a control unit 72, a local oscillation unit 73, a mixing unit 74, an HPF (high pass filter) 75, a gain adjustment unit 76, and an addition unit 77. And is made up of. The operation of the high frequency signal adding unit 61 will be described later.

  The listening mode processing unit 62 processes the signal output from the high frequency signal adding unit 61, performs signal processing corresponding to a listening mode such as monaural, stereo, and surround, and outputs the processed signal to the amplifier unit 15.

  The amplifier unit 15 includes a D / A converter and an analog amplifier (not shown), converts the data input from the audio DSP 14 into an analog signal by D / A conversion, amplifies the signal, and supplies the amplified signal to the speaker unit 16. .

  The speaker unit 16 converts the audio signal supplied from the amplifier unit 15 into air vibration and outputs audio.

  The key control unit 17 receives a user operation input using an operation input key such as a remote controller and supplies data corresponding to the input signal to the system microcomputer 11.

  The PC 18 includes a storage device such as a hard disk device, and supplies content data stored in the storage device to the client control IC 12 via the router 19 and the LAN 22 in response to a request from the client control IC 12.

  The router 19 is a device for relaying data exchange between the PC 18 and the client control IC 12 via the LAN 22.

  The memory card 20 is a recording medium that incorporates a nonvolatile semiconductor memory such as a flash memory. The memory card 20 is inserted into a slot 21 connected to the client control IC 12 and supplies content data recorded inside the memory card 20 itself to the client control IC 12 via the slot 21.

  The slot 21 is an insertion slot for the memory card 20 and is connected to the client control IC 12. The slot 21 reads the content data recorded on the memory card 20 and supplies it to the client control IC 12.

  The LAN 22 is composed of Ethernet (registered trademark) or the like, and mediates exchange of data between the client control IC 12 and the PC 18.

  The LAN controller 23 is a network control device for controlling exchange of data between the client control IC 12 and the PC 18 via the LAN 22, and is connected to the client control IC 12.

  The display unit 24 is disposed on, for example, a front panel of the content reproduction apparatus, is configured of a liquid crystal display panel, a fluorescent display tube, and the like, and according to display data supplied from the display control microcomputer 25, the operation status of the AV device 1 A message indicating the like is displayed.

  The display control microcomputer 25 includes a CPU, a ROM, a RAM, and the like, and controls display contents displayed on the display unit 24 under the control of the system microcomputer 11.

  The LAN cable connection terminal 26 connects the LAN 22 and the LAN controller 23.

Next, the operation of the audio system having the above configuration will be described.
In response to an instruction from the key control unit 17, the system microcomputer 11 supplies audio data supplied from the PC 18 to the client control IC 12 via the LAN 22 and the LAN controller 23 and from the memory card 20 via the slot 21. Which audio data to select is selected.

  In accordance with the instruction, the system control unit 53 of the client control IC 12 supplies audio data supplied from the PC 18 to the selector unit 51 via the LAN 22 and the LAN controller 23 and audio supplied from the memory card 20 via the slot 21. Lets you select one of the data.

  The system control unit 53 of the client control IC 12 transmits attribute information such as a file name supplied from the selected source to the system microcomputer 11. The system microcomputer 11 provides these information to the display control microcomputer 25. The display control microcomputer 25 displays the information on the display unit 24 as appropriate.

  Subsequently, when playback (PLAY) is instructed from the key control unit 17, the system microcomputer 11 instructs the system control unit 53 to perform playback.

The system control unit 53 reads data from the selected file of the selected source in accordance with the instruction.
This file data includes audio data encoded or compression encoded by a predetermined encoding method, control information for specifying the file format, sampling frequency, bit rate, etc., file name, album name, player name, performance Text information such as time.

The system control unit 53 determines the encoding method or compression format of the file from the control information, for example, the extension of the file name and the bit rate, and designates the decoding method to the decoding DSP 52.
The decoding unit 52 decodes the audio data supplied from the selector 51, converts it into a PCM signal, and supplies it to the audio DSP 14 via the DIR 13.

  Further, the system control unit 53 provides the control information and text information of the audio data being reproduced to the system microcomputer 11. The system microcomputer 11 provides text information to the display control microcomputer 25 and causes the display unit 24 to display the song name and the like.

The system microcomputer 11 notifies the control information to the high frequency signal adding unit 61 of the audio DSP 14.
The high-frequency signal adding unit 61 generates a high-frequency signal that is missing from the encoded / compressed audio data stored in the source according to the control information, and generates the audio signal reproduced from the source. Adds a more natural sound field.

  In general, an audio signal represented by input PCM data based on a compression-coded audio signal is a component having a frequency equal to or higher than a certain value in the original audio, as shown in FIG. 4A, for example. Has a spectral distribution corresponding to that from which the is removed.

  This constant value (the value indicated as “fIN” in FIG. 4A) indicating the upper limit of the occupied band of the audio signal represented by the PCM data input from the DIR 13 is, for example, input audio data in MP3 format and a bit rate of 128 kbps. When it is made up of data, it is about 16 kilohertz, and when it is made up of ATRAC3 format data, it is about 14 kilohertz.

  PCM data from the DIR 13 is supplied to the delay unit 71 and the mixing unit 74, and control data from the system microcomputer 11 is supplied to the control unit 72.

  When the PCM data is supplied, the delay unit 71 delays the PCM data and supplies the delayed data to the adder 77.

  The delay time of the delay unit 71 is equal to the length of time that elapses until the signal supplied to the mixing unit 74 is supplied to the adding unit 77 after being processed by the mixing unit 74, the HPF 75 and the gain adjusting unit 76. Further, the phase of the delayed signal (analog signal represented by the PCM data) supplied from the delay unit 71 to the adder 77 and the signal (analog represented by the PCM data) supplied from the gain adjustment unit 76 to the adder 77. The phase of the signal) is substantially in phase between the signals supplied to the adder 77 at the same time.

  The control unit 72 determines the frequency of the local oscillation signal generated by the local oscillation unit 73 based on the control information including the file format and / or the bit rate information, and supplies the local oscillation unit 73 with information indicating the determined frequency. To do. In addition, the control unit 72 determines the passband characteristic of the HPF 75 based on the control information, and supplies the HPF 75 with information indicating the determined passband characteristic.

  Specifically, the control unit 72 further includes, for example, a nonvolatile memory such as an EEPROM (Electrically Erasable / Programmable Read Only Memory), and the nonvolatile memory stores a bandwidth table and a Q value table in advance. .

  The bandwidth table is a table for storing data indicating the upper limit value of the occupied bandwidth of the input audio signal for each of the encoding method and the compression method. The Q value table is a table that stores data that specifies the Q value of the HPF 75 for each genre of music represented by the input audio data. The value of Q is in the range from about 1 to about 7.

  And the control part 72 performs the process specifically described as (1)-(4) below.

(1) First, the control unit 72 specifies the upper limit value of the occupied bandwidth of the audio data in the format indicated by the control information by searching the bandwidth table.

(2) Next, the control unit 72 determines the passband of the HPF 75 and the frequency of the local oscillation signal so as to meet the following conditions (a) and (b). That is, as shown in FIGS. 4B and 4C,
(A) The frequency fOSC of the local oscillation signal is lower than the upper limit frequency fIN of the occupied band of the input audio signal,
(B) The frequency fP of the gain peak of the HPF 75 is lower than the above-described frequency fIN and higher than the frequency (fIN−fOSC).
Thus, the pass band of the HPF 75 and the frequency of the local oscillation signal are determined so that the relationship is established. That is, there is a relationship of the following expressions between the values of fIN, fOSC, and fP.
fOSC <fIN
(FIN−fOSC) <fP <fIN

  However, a signal representing music generally has a characteristic that a component of 10 kHz or more has a spectrum distribution including a large number of peaks arranged at intervals of about several hundred hertz while being attenuated as the frequency increases.

  For this reason, in order to add such a characteristic component to the input audio signal and generate an output audio signal by the processing described later, the frequency of the audio signal is about 10 as shown in FIG. 4B. It is desirable that the component of kilohertz or higher (or the component having the characteristic even if not higher than 10 kilohertz) is frequency-converted by the processing of the mixing unit 74 so as to occupy a band of frequency fIN or higher. In order to obtain the result of such frequency conversion, for example, if the portion having the characteristic in the input signal is a frequency component of about 10 kHz or more, the frequency fOSC of the local oscillation signal is set to about (fIN−10 kHz). The

  (3) While performing the processing of (2), the control unit 72 searches the Q value table using the music genre indicated by the control information as a search key, thereby determining the Q value that the HPF 75 should take.

  (4) Then, the control unit 72 supplies the local oscillation unit 73 with information indicating the value of fOSC that satisfies the above condition (a). Also, information specifying the value of fP satisfying the above condition (b) as the peak frequency of the passband characteristic of the HPF 75, and information specifying the Q value determined in the process (3) above, Supply to HPF75.

  When the information indicating the frequency of the local oscillation signal is supplied from the control unit 72, the local oscillation unit 73 generates a local oscillation signal having the frequency indicated by this information and supplies it to the mixing unit 74. The local oscillating unit 73 is composed of, for example, an IIR (Infinite Impulse Response) type low-pass filter. However, it is assumed that the Q value of the low-pass filter is set to practically infinite.

  The local oscillation unit 73 generates a local oscillation signal that is substantially a sine wave.

  The mixing unit 74 is supplied with the same signal as that supplied to the delay unit 71 at the same time as the delay unit 71. Then, by mixing this signal and the local oscillation signal generated by the local oscillation unit 73, a signal representing the product of the input audio signal and the local oscillation signal is generated, and the generated signal is supplied to the HPF 75.

  The signal supplied from the mixing unit 74 to the HPF 75 includes a component (sum component) having a frequency corresponding to the sum of the frequency of the input audio signal and the frequency of the local oscillation signal, and a frequency corresponding to the difference between the frequency of the input audio signal and the local oscillation signal. The component which has (difference component) is included. The spectrum of the sum component occupies a band with the frequency fOSC as the lower limit and the frequency (fIN + fOSC) as the upper limit, as shown in FIG. Further, the spectrum of the difference component occupies a band whose upper limit is the frequency (fIN−fOSC), as shown in FIG.

  The HPF 75 filters the component supplied from the mixing unit 74 and supplies the filtered component to the gain adjusting unit 76. As shown in FIG. 5, the HPF 75 has a passband characteristic corresponding to an IIR type high-pass filter, as shown in FIG. 5, and has a peak, and the frequency increases on the higher frequency side than this peak. As a result, the attenuation rate of the HPF 75 increases. Then, the HPF 75 adjusts the peak frequency and Q value to values indicated by the information supplied from the control unit 72. The larger the value of Q, the steeper the peak of the passband characteristic of the HPF 75.

  The sum component and difference component of the signal supplied from the mixing unit 74 to the HPF 75 occupy a band as shown in FIG. On the other hand, the peak frequency fP of the passband characteristic of the HPF 75 exceeds (fIN−fOSC) and is less than fIN. Accordingly, as shown in FIG. 4C, the HPF 75 passes a sum component having a frequency not less than fP and not more than fIN + fOSC among signals supplied from the mixing unit 74, and substantially blocks other components. However, as described above, the HPF 75 has a higher attenuation factor as the frequency increases on the higher frequency side than the peak of the passband characteristics. Therefore, the sum component that passes through the HPF 75 also has a higher attenuation as the frequency increases.

  The gain adjusting unit 76 amplifies the signal supplied from the HPF 75 and supplies the amplified signal to the adding unit 77. The gain of the gain adjusting unit 76 is set to such a value that a spectrum envelope of an output signal described later generated by the adding unit 77 becomes smooth in the vicinity of the frequency fIN. This value is set based on, for example, experiments.

  The adder 77 generates and outputs a signal representing the sum of the delayed signal supplied from the delay unit 71 and the signal supplied from the gain adjuster 76.

  As shown in FIG. 4D, the output signal is composed of a component corresponding to the input signal and an added component. The added component is frequency-converted to the high frequency side by an amount corresponding to the difference between the predetermined value and the upper limit frequency of the occupied band of the input audio signal, in the input signal. Is obtained.

  When the input audio signal is a band-limited signal, the component removed from the original audio signal is likely to be composed of higher harmonic components of components of 10 kilohertz or more in the original audio signal. Therefore, when the input audio signal is a signal whose band is limited, the output audio signal is close to the original audio signal before the band is limited.

  The component added to the input audio signal in the output audio signal is a component on the higher frequency side than the peak of the passband characteristic of the HPF 75, and the higher the frequency, the greater the attenuation. For this reason, the spectrum of the output sound signal shows a natural distribution close to the typical spectrum of sound, such that the intensity decreases as the frequency of the component increases.

  In addition, the frequency of the local oscillation signal and the passband characteristic of the HPF 75 change according to the occupied bandwidth of the input signal that the control information represents directly or indirectly, so that the sound suitable for the input signal having various occupied bandwidths can be obtained. Field interpolation can be performed.

  Further, by changing the Q value of the HPF 75 according to the genre information included in the control information, the shape of the envelope of the high frequency component added to the input audio signal in the output signal changes. Using this, a spectrum distribution suitable for the music genre is given. Specifically, in the case of a lock that frequently uses high-frequency sound sources such as cymbals, the control unit 72 lowers the Q value so that many high-frequency components remain without being attenuated. In the case of a classic that is not performed, the high frequency component is greatly attenuated by increasing the Q value. When an encoded signal that does not need to be added with an interpolation signal is input, such as when the value of the upper limit frequency of the occupied band specified by the above processing indicates 22 kHz, the above interpolation processing is performed. It is controlled not to be executed. The control is performed by the control unit 72 based on the received control information. If interpolation processing is not performed, only the signal path that passes through the delay unit 71 among the PCM signals input from the DIR 13 is enabled and the path that passes through the mixing unit 74 is substantially cut off. Good. As a signal that is not subjected to interpolation, for example, an MP3 format encoding format, which is encoded at a bit rate of 256 kbps, corresponds. Of course, there are various other combinations of encoding formats and bit rates, so whether or not to perform the interpolation processing is determined corresponding to each. The criterion for determining whether or not to perform the interpolation processing may be selected as appropriate, but as a guideline, if there is an audio signal in a frequency band of 20 kHz or higher, the interpolation processing may not be performed.

  Note that the control information may directly indicate the value of the upper limit frequency of the occupied band in accordance with the encoding method of the audio data being reproduced.

  In the above, in order to facilitate understanding, the contents of signal processing have been described based on the functions realized by the DSP, but in reality, they are realized by sequentially processing digital data in accordance with a predetermined program described below. Is done.

  First, signals input from the DIR 13 are sequentially stored in the buffer. Based on the control information loaded from the system microcomputer 11, the frequency of the local oscillation signal and the passband characteristics to be obtained by filtering in step S5 described later are determined (step S1). At this point, since the upper limit frequency included in the audio signal can be specified from the control information, it is determined whether or not to perform the subsequent interpolation process. In the case of an audio signal that does not need to be interpolated and the audio signal exists up to a high frequency, the following interpolation processing is not performed.

  One sample is acquired from the input data (step S2). Further, data for one sample of the local oscillation signal having the frequency determined in step S1 is generated (step S3). In step S2, the first one sample that has not been acquired in step S2 is acquired, and in step S3, the data whose phase is delayed by one sample from the last data generated in step S3 (however, In the first step S3, data having an arbitrary initial phase is generated.

  Next, the product of the data value acquired in step S2 and the sample value of the local oscillation signal generated in step S3 is calculated (step S4), and the obtained product is filtered (step S5).

  Then, the value of the data acquired in step S2 and the value obtained as a result of filtering in step S5 are added (step S6), data indicating the addition result is output (step S7), and the process returns to step S2. .

  By such processing, the high frequency signal adding unit 61 outputs a signal in which the high frequency band component missing in the encoded audio data stored in the PC 18 or the memory card 20 is complemented.

  The listening mode processing unit 62 of the audio DSP 14 processes the input audio signal according to the listening mode (monaural, stereo, surround, etc.) designated by the system microcomputer 11 and emits sound from the speaker unit 16 via the amplifier unit 15. To do.

  With such a configuration, control information that can specify the encoding / compression method of the audio signal to be played back acquired by the client control IC 12 is transmitted to the system microcomputer 11 together with the text information for display, and the like. It can be transmitted to the audio DSP 14 via a route for designating other control information (information for designating the listening mode) and the like, and the audio DSP 14 can perform high-frequency range complementation processing. This makes it possible to transmit necessary control information while distributing the processing load.

  In addition, this invention is not limited to what was shown by the said embodiment, A various deformation | transformation and application are possible. For example, the high-frequency range complementing method itself is arbitrary. Also, the type and content of the control information are arbitrary.

  Also, the type of source for supplying the content is not limited to the PC 18 (HDD) or the memory card 20, but may be, for example, a DVD (Digital Versatile Disc), a CD (Compact Disc), or the like. In this case, for example, an output signal from the source is supplied to the client control IC 12 when decoding is necessary, and is supplied to the DIR 13 when decoded, and an arbitrary input is selected by the DIR 13.

1 is a configuration diagram of an AV system according to an embodiment of the present invention. FIG. 2 is a block diagram showing in detail a configuration of a part related to sound field complementing processing in the AV system shown in FIG. 1. It is a block diagram which shows the function of the sound field complementation part of an audio DSP. It is a spectrum figure for demonstrating a sound field complementation process. It is a graph of an example of the pass-band characteristic of the high pass filter shown in FIG. It is a flowchart of the sound field complementation process shown in FIG. It is a block diagram of the conventional AV system.

Explanation of symbols

1 AV device 11 System microcomputer 12 Client control IC
13 DIR
14 Audio DSP
15 Amplifier unit 16 Speaker unit 17 Key control unit 18 PC
19 router 20 memory card 21 slot 22 LAN
23 LAN controller 24 Display unit 25 Display control microcomputer 26 LAN cable connection terminal

Claims (4)

Decoding means for decoding and outputting the encoded audio data and obtaining information relating to the audio data, reproduction means for processing the decoded audio data and reproducing an audio signal, and audio data from the decoding means Control means for receiving information and controlling the operation of the playback means, and an audio device comprising:
The decoding means supplies information corresponding to the encoding method of the audio data to the control means,
The control means supplies information corresponding to the information corresponding to the encoding method from the decoding means to the reproducing means,
The reproduction unit includes a complementing unit that executes a sound field complementing process that complements a signal in a frequency band that is missing in the encoding scheme, based on information from the control unit.
An audio device characterized by that. Any of the decoding means, the control means, and the reproduction means includes means for obtaining a control parameter for sound field interpolation processing corresponding to the encoding method,
The reproducing means includes a complementing means for executing a sound field complementing process for complementing a signal in a frequency band corresponding to the encoding method according to a control parameter.
The audio apparatus according to claim 1. The interpolation means includes
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal that represents a speech that is subject to spectrum interpolation;
A filter that extracts a component in a second band on a higher frequency side than the first band occupied by the interpolated signal in the interpolation signal;
An adder for generating an output signal representing the sum of the interpolated signal and the component extracted by the filter;
The audio apparatus according to claim 1, further comprising: The information corresponding to the encoding method includes a file format and / or a bit rate.
The audio device according to any one of claims 1 to 3, wherein

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