JP2007311965A - Digital audio signal processor - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To generate a sound signal of high quality by preventing distortion of the signal due to clipping when the sound signal having the predetermined number of channels is generated based upon an input source signal, and to improve sounding and articulation of a reproduced sound during small-sound-volume reproduction. <P>SOLUTION: A digital audio signal processor 1 which generates the sound signal having the predetermined number of channels based upon the input source signal S0 has a processing section 10 which controls the level of a sound signal of each channel according to the number of channels of the input source signal S0 to emphasize both or either of low-frequency and high-frequency levels of the sound signal, so even when input source signals S0 having various numbers of channels are input, distortion of signals due to clipping is prevented to generate sound signals S1, SAL, and SAR of high quality. The sounding and articulation of the reproduced sound during small-sound-volume reproduction is improved. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

  The present invention relates to a digital audio signal processing apparatus.

  In general, multi-channel playback devices such as AV (Audio Visual, Audio Video) amplifiers and DVD (Digital Versatile Disc) players use multi-channel digital audio recorded on a multi-channel DVD or CD (Compact disk). The left and right front, center, left and right surround, and in addition to these, the source (sound) signal of each channel of the subwoofer is generated, and five or more numbers are arranged around the listener. By outputting to a speaker, a three-dimensional sound field full of realism is formed.

  On the other hand, when listening to the three-dimensional sound reproduced by such a multi-channel playback device through two speakers such as headphones, a multi-channel source signal is synthesized to generate a two-channel (2ch) sound signal. There is a need. The synthesis of a multi-channel source signal into a 2-channel sound signal in this way is called “stereo downmix”. By reproducing the 2ch sound signal obtained by the stereo downmix with the left and right speakers, the listener can hear the realistic 3D sound similar to the surround system. Such headphones are called “surround headphones”.

  Note that converting a multi-channel source signal into a source signal having a smaller number of channels is called “virtual surround decoding”, and the resulting signal is called “virtual surround signal”. The stereo downmix is an example of a bar channel surround decoding.

  For example, Patent Literature 1 describes an example of the stereo downmix and the surround headphones.

JP 2003-333700 A

  By the way, when listening to content such as music, movies, television, etc. at night or in a quiet environment, if you set the volume level of the playback sound to be relatively low considering the surroundings, it will be difficult to hear minute sounds. There are cases where problems such as sound reverberation and realistic sensation are reduced as compared with listening at a louder volume than a predetermined level. As a specific example for solving this problem, for example, a technique is known in which the high-frequency component and low-frequency component of reproduced sound are enhanced to improve the sound reverberation and serif clarity.

  However, if the high-frequency component and the low-frequency component are simply emphasized in the 2ch acoustic signal after the stereo downmix described above, the level of the emphasized audio data exceeds the maximum sound volume level of 0 dB (so-called clipping). May cause distortion in the signal and lower the sound quality.

  By the way, as input source signals (for example, digital audio data), for example, 5.1ch (5ch and subwoofer channel (LFE: Low Frequency Effect)), 2ch (stereo), 1ch (monaural), other 6.1ch, and more. Source signals of various channels such as a plurality of channels. This subwoofer (LFE) is used to add force to the sound by outputting a heavy bass in a band of about 20 Hz to 120 Hz as an auxiliary output. For example, after synthesizing the source signals of these various channel numbers to generate a 2-channel acoustic signal, the low-frequency component and high-frequency component are simply emphasized for the 2-channel acoustic signal regardless of the number of channels of the source signal. Then, the volume level of the 2ch acoustic signal after synthesis differs depending on the number of channels of the input source signal. Specifically, for example, as the number of channels of the input source signal increases, the level of the 2ch acoustic signal after synthesis increases. In some cases, the level of the audio data exceeds 0 dB, which is the maximum volume level (clipping), and the signal is distorted to deteriorate the sound quality.

  This invention makes it an example of a subject to cope with such a problem. In other words, when generating an acoustic signal with the specified number of channels based on the input source signal, it prevents the distortion of the signal due to clipping and generates a high-quality acoustic signal, and the sound of the reproduced sound during small volume playback. It is an object of the present invention to improve clarity and the like.

  In order to achieve such an object, the present invention comprises at least the configurations according to the following independent claims.

  The invention according to claim 1 is a digital audio signal processing apparatus that generates an acoustic signal of a prescribed number of channels based on an input source signal, wherein the acoustic signal of each channel is generated according to the number of channels of the input source signal. It has a processing part which controls the level of a signal and emphasizes the level of either or both of the low frequency and high frequency of the acoustic signal.

  The invention according to claim 2 is a digital audio signal processing apparatus that generates acoustic signals of a prescribed number of channels based on an input source signal, wherein the input source signal has a format in which a dynamic range can be controlled. And controlling the dynamic range to emphasize both or one of the low frequency and high frequency levels of the acoustic signal with a first amplification factor, and when the input source signal is in another format, the acoustic signal A processing unit that emphasizes both or one of the low frequency and high frequency levels with a second amplification factor larger than the first amplification factor.

  According to a third aspect of the present invention, a multi-channel decoder unit that generates a first acoustic signal having a number of channels corresponding to the number of channels of the input source signal based on the input source signal, and the first acoustic signal A digital audio signal processing apparatus having an audio signal conversion unit for generating a second audio signal having a predetermined number of channels based on the level of the first audio signal generated by the multi-channel decoder unit A correction unit that attenuates at a specified attenuation rate, and a low frequency and / or a high frequency level of the second acoustic signal generated by the acoustic signal conversion unit at a specified amplification rate An amplification unit that amplifies, and a control unit that controls the attenuation rate of the correction unit and the amplification rate of the enhancement unit based on the number of channels of the first acoustic signal. That.

  According to a fifth aspect of the present invention, a multi-channel decoder unit that generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal based on the input source signal, and the first acoustic signal A digital audio signal processing device having an acoustic signal conversion unit that generates a second acoustic signal having a predetermined number of channels based on a correction unit that controls a dynamic range of the first acoustic signal; Based on the format information of the input source signal and the emphasis unit that amplifies the low frequency and / or high frequency level of the second acoustic signal generated by the acoustic signal conversion unit at a specified amplification factor When it is determined that the dynamic range of the first acoustic signal can be controlled, the correction unit dynamically adjusts the dynamic range of the first acoustic signal. And the amplification unit amplifies at a first amplification factor, otherwise the control by the correction unit is suppressed and the enhancement unit has a second amplification larger than the first amplification factor. And a control unit that amplifies at a rate.

  A digital audio signal processing apparatus according to an embodiment of the present invention is a digital audio signal processing apparatus that generates an acoustic signal having a specified number of channels based on an input source signal, and that corresponds to the number of channels of the input source signal. And a processing unit that controls the level of the acoustic signal of each channel and emphasizes the level of one or both of the low and high frequencies of the acoustic signal.

  In the digital audio signal processing apparatus having the above configuration, the processing unit controls the level of the acoustic signal of each channel according to the number of channels of the input source signal, and both or one level of the low frequency and the high frequency of the acoustic signal. Therefore, even when input source signals having various numbers of channels are input, distortion of the signal due to clipping can be prevented and a high-quality acoustic signal can be generated. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  In addition, a digital audio signal processing apparatus according to an embodiment of the present invention includes a multi-channel decoder unit that generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal based on the input source signal. A digital audio signal processing device having an acoustic signal conversion unit that generates a second acoustic signal having a predetermined number of channels based on the first acoustic signal, the digital audio signal processing device generating a second acoustic signal generated by the multi-channel decoder unit; A correction unit that attenuates the level of one acoustic signal at a prescribed attenuation rate, and a low-frequency and / or high-frequency level of the second acoustic signal generated by the acoustic signal conversion unit are defined. An emphasis unit that amplifies at an amplification factor, and a control unit that controls the attenuation rate of the correction unit and the amplification factor of the enhancement unit based on the number of channels of the first acoustic signal And wherein the door.

  In the digital audio signal processing apparatus having the above configuration, the control unit controls the attenuation rate of the correction unit and the amplification rate of the enhancement unit based on the number of channels of the first acoustic signal. Even when the first acoustic signal with the number of channels is generated with the specified number of channels, for example, the second acoustic signal with two channels, the distortion of the signal due to clipping is prevented and a high-quality acoustic signal is generated. can do. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  A digital audio signal processing apparatus according to an embodiment of the present invention is a digital audio signal processing apparatus that generates acoustic signals of a specified number of channels based on an input source signal, and the input source signal has a dynamic range. In the case of a format that can be controlled, the dynamic range is controlled to emphasize the low and / or high frequency levels of the acoustic signal with the first amplification factor, and the input source signal has the other format. In some cases, a processing unit that emphasizes both or one of the low frequency and high frequency levels of the acoustic signal with a second amplification factor larger than the first amplification factor is provided.

  In the digital audio signal processing apparatus having the above-described configuration, when the input source signal has a format in which the dynamic range can be controlled, the processing unit controls the dynamic range so that the low and / or high frequency levels of the acoustic signal are controlled. Emphasizing by the first amplification factor, when the input source signal is in other formats, the low and / or high frequency levels of the acoustic signal are set to a second amplification factor larger than the first amplification factor. Therefore, if the input source signal is in a format that can control the dynamic range, even if it is in any other format, a higher-quality acoustic signal is generated by performing optimum enhancement processing for each format. be able to. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  In addition, a digital audio signal processing apparatus according to an embodiment of the present invention includes a multi-channel decoder unit that generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal based on the input source signal. A digital audio signal processing apparatus having an acoustic signal conversion unit that generates a second acoustic signal having a predetermined number of channels based on the first acoustic signal, wherein the dynamic range of the first acoustic signal is reduced. A correction unit for controlling, an emphasis unit for amplifying the low-frequency and / or high-frequency levels of the second acoustic signal generated by the acoustic signal conversion unit at a specified amplification factor, and an input source signal When it is determined that the dynamic range of the first acoustic signal can be controlled based on the format information, the correction unit can detect the first acoustic signal. In addition to controlling the dynamic range, the enhancement unit amplifies at the first amplification factor, and in other cases, the control by the correction unit is suppressed and the enhancement unit has a second amplification factor greater than the first amplification factor. And a control unit for amplification.

  In the digital audio signal processing device configured as described above, when the control unit determines that the dynamic range of the first acoustic signal can be controlled based on the format information of the input source signal, the correction unit performs the first operation. The dynamic range of one acoustic signal is controlled and the enhancement unit amplifies at the first amplification factor. In other cases, the control by the correction unit is suppressed and the enhancement unit has a second higher amplification factor than the first amplification factor. Therefore, if the input source signal is in a format that can control the dynamic range, even if the input source signal is in any other format, it is possible to achieve higher quality sound by performing optimum enhancement processing for each. A signal can be generated. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  A digital audio signal processing apparatus according to an embodiment of the present invention will be described below with reference to the drawings.

[First Embodiment]
FIG. 1 is a functional block diagram for explaining an audio system 100 employing a digital audio signal processing apparatus 1 according to the first embodiment of the present invention.

  In the audio system 100 according to the present embodiment, the digital audio signal processing apparatus 1 is capable of reproducing digital audio data recorded in a multi-channel on a storage medium such as a DVD, CD, hard disk, and semiconductor memory. The source signal (digital audio data) S0 having various channel numbers input via a wired or wireless communication channel is subjected to a predetermined process according to the present invention to provide a predetermined number of channels, for example, 2ch acoustic signals. (Digital signal) S1 (L02, R02) is generated. The acoustic signal S1 (L02, R02) is converted into an analog signal SAL, SAR by, for example, a digital / analog converter (DAC) 16, amplified by an amplifier 3 (3L, 3R), and produced by a speaker 4 (4L, 4R). Is done. The amplification amount of the amplifier 3 can be set by a user operation, for example.

  The source signal S0 corresponds to an embodiment of an input source signal according to the present invention, and the acoustic signal S1 corresponds to an embodiment of an acoustic signal having a defined number of channels according to the present invention.

  As the source signal S0 (multichannel audio data) input to the digital audio signal processing device 1, for example, 5.1ch (left front signal L, right front signal R, center signal C, left surround signal LS, right surround signal RS) , Subwoofer signal LFE), 2ch (left front signal L, right front signal), 1ch (center signal C), and the like. In addition, as the source signal S0, for example, 6.1ch (left front signal L, right front signal R, center signal C, left surround signal LS, right surround signal RS, center surround signal CS, subwoofer signal LFE). Or a source signal of more than one channel.

  As shown in FIG. 1, the communication unit 19 is provided in the digital audio signal processing device 1, and the receiving side device 200 generates a sound from the speaker 204 based on the acoustic signal S <b> 1 (L02, R02) transmitted from the communication unit 19. Form may be sufficient. For example, as illustrated in FIG. 1, the reception-side device 200 includes a communication unit 201 that performs data communication with the communication unit 19 of the digital audio signal processing device 1 by a wireless method or a wired method, and an acoustic signal S1 ( L02, R02), a digital / analog converter (DAC) 202 that performs digital / analog conversion, an amplifier 203 (203L, 203R) that amplifies the analog signal generated by the DAC 202, and generates sound in response to the analog signal from the amplifier 203 And a speaker 204 (204L, 204R). The audio system 100 can be applied to various audio systems such as wireless headphones, an in-vehicle audio system, an indoor audio system, and an outdoor audio system.

  The digital audio signal processing apparatus 1 according to the present embodiment, for example, when listening to content such as music, movies, and televisions at night or in a quiet environment, has a relatively low volume level of reproduced sound in consideration of the surroundings. When set, according to the present invention, according to the number of channels of the input source signal according to the present invention, the level of the acoustic signal of each channel is controlled to generate an acoustic signal of a predetermined number of channels, and the level of the generated acoustic signal is increased. By emphasizing the low-frequency component and the low-frequency component, the sound and the clarity of the speech are improved.

  Hereinafter, the digital audio signal processing apparatus 1 will be described in detail with reference to the drawings.

[Digital audio signal processing apparatus 1]
When the digital audio signal processing apparatus 1 according to the present embodiment generates the acoustic signal having the number of channels defined based on the input source signal S0, the digital audio signal processing apparatus 1 generates the acoustic signal of each channel according to the number of channels of the input source signal S0. A processing unit 10 that controls the level and emphasizes the level of one or both of the low and high frequencies of the acoustic signal is included.

  Specifically, as shown in FIG. 1, the digital audio signal processing apparatus 1 includes a digital audio interface receiver (DIR) 11, a multichannel decoder unit (decoder unit: MCD) 12, a correction unit 30, and a stereo downmix unit. (Downmix section: SDMP) 13, digital audio data high frequency enhancement section (high frequency enhancement section: TB (Treble boost)) 14, digital audio data low frequency enhancement section (low frequency enhancement section: BB (Bass boost)) 15 A digital / analog converter (DAC) 16 and a control circuit (MP: microprocessor) 17.

  The multi-channel decoder unit 12 corresponds to one embodiment of the multi-channel decoder unit according to the present invention, and the stereo downmix unit 13 corresponds to one embodiment of the acoustic signal conversion unit according to the present invention. The correction unit 30 corresponds to an embodiment of the correction unit according to the present invention, and the enhancement unit 40 including the high frequency enhancement unit 14 and the low frequency enhancement unit 15 corresponds to an embodiment of the enhancement unit according to the present invention. The control circuit 17 corresponds to an embodiment of the control unit according to the present invention. For example, the processing unit 10 includes a DIR 11, a decoder unit 12, a correction unit 30, a downmix unit 13, an enhancement unit 40 (a high frequency enhancement unit 14 and a low frequency enhancement unit 15), and a control circuit 17. Corresponds to an embodiment of the processing unit according to the present invention.

  In the present embodiment, the decoder unit 12, the correction unit 30, the downmix unit 13, the high frequency emphasizing unit 14, and the low frequency emphasizing unit 15 are related to the present invention by a DSP (Digital Signal Processor) executing a program. The function is realized, but is not limited to this form. For example, the decoder unit 12, the correction unit 30, the downmix unit 13, the high frequency emphasizing unit 14, the low frequency emphasizing unit 15, and all or part of the control circuit 17 may be realized by a DSP or a dedicated circuit. . Further, the functions according to the present invention may be realized by a general-purpose processor executing a program stored in a memory.

  The DIR 11 extracts, for example, a clock signal (CK) and data (DATA) from a source signal (digital audio data) S0 input from the audio device 2 and demodulates the demodulated clock signal (CK) and data DT. 12 is output.

  The DIR 11 outputs the data signal (DATA) only to the decoder unit 12, but is not limited to this form. For example, the DIR 11 may output a data signal (DATA) to the control circuit 17.

  Based on the input source signal (digital audio data) S0, the decoder unit (MCD) 12 generates an acoustic signal S12 having the number of channels corresponding to the number of channels of the input source signal. For example, the decoder unit 12 generates an acoustic signal S12 from the data DT input from the DIR 11 according to the clock signal (CK), and outputs the acoustic signal S12 to the downmix unit 13.

  For example, the decoder unit 12 receives the data DT of 5.1 ch (more specifically, 6 channels of 5 channels and subwoofer channels), the left front signal L, the right front signal R, the center signal C, the left surround. When 6-ch acoustic signal S12 of signal LS, right surround signal RS, and subwoofer signal LFE is generated and 2ch (stereo) data DT is input, 2ch acoustic signal of left front signal L and right front signal R is input. When S12 is generated and 1ch (monaural) data DT is input, a 1ch acoustic signal S12 of the center signal C is generated.

  Further, the decoder unit 12 outputs channel number data SD1 indicating the number of channels of the acoustic signal S12 to the control circuit 17. At this time, for example, the decoder unit 12 may output data indicating the number of channels of audio data included in the source signal S10 or data indicating the number of channels included in the data DT by the DIR 11 as the channel number data SD1.

  In addition, the decoder unit 12 may generate the acoustic signal S12 having a larger number of channels than the number of channels of the source signal S0 by a predefined surround process. Specifically, for example, when 2ch (stereo) data DT is input, the decoder unit 12 expands the 2ch audio data by performing a predefined surround process, and thus 5.1ch (details). 6ch) acoustic signal S12 may be generated. Alternatively, 1ch audio data may be expanded to generate a 2ch or 5.1ch (specifically 6ch) acoustic signal S12.

  Further, the decoder unit 12 may generate the acoustic signal S12 having a smaller number of channels than the number of channels of the source signal S0 through a surround process defined in advance. For example, the decoder unit 12 may generate the 5.1ch acoustic signal S12 from the 10-channel audio data DT, or may generate the 2ch acoustic signal S12 from the 5.1ch (6ch) channel audio data DT. Good.

  When the number of channels of the original source signal S0 and the number of channels of the generated acoustic signal S12 are different, the decoder unit 12 sets, for example, the number of channels of the generated acoustic signal S12 to the control circuit 17 as channel number data SD1. Output. That is, the decoder unit 12 outputs the number of channels of the signal S12 output to the downmix unit 13 to the control circuit 17 as channel number data SD1.

  Then, the control circuit 17 acquires the number of channels of the acoustic signal S12 generated by the decoder unit 12 based on the input channel number data SD1.

  The acquisition method of the number of channels of the acoustic signal S12 by the control circuit 17 is not limited to the above-described form. For example, when the channel number data SD1 indicating the number of channels of the source signal S0 (data DT) is input, the control circuit 17 associates the number of channels of the source signal S0 (data DT) with the number of channels of the acoustic signal S12. Based on the association data, the number of channels of the acoustic signal S12 generated by the decoder unit 12 may be determined.

  The downmix unit (SDMP) 13 is based on the multi-channel or single-channel acoustic signal S12 generated by the decoder unit 12 (in detail, based on the acoustic signal S12 corrected by the correcting unit 30 described later). The acoustic signal S13 having a predetermined number of channels is generated. The downmix unit 13 according to the present embodiment generates, for example, a 2ch acoustic signal S13 (L0, R0) by synthesizing the acoustic signal S12 of each channel based on a preset downmix coefficient, The acoustic signal S13 (L0, R0) is output to the enhancement unit 40.

  Hereinafter, a case where a 5.1ch (specifically 6ch) acoustic signal S12 is input to the downmix unit 13 will be described.

  FIG. 2 is a circuit diagram for explaining the downmix unit 13 according to the embodiment of the digital audio signal processing apparatus 1 shown in FIG. The downmix unit 13 includes a left channel synthesis unit 310 and a right channel synthesis unit 320, for example, as shown in FIG. The left channel combining unit 310 includes a plurality of multipliers 311 to 316 and an adder 317. The right channel combining unit 320 includes a plurality of multipliers 321 to 326 and an adder 327. The left channel synthesizing unit 310 and the right channel synthesizing unit 320 having the above-described configuration are defined in advance with a left front signal L, a right front signal R, a center signal C, a left surround signal Ls, a right surround signal Rs, and a subwoofer signal LFE. The downmix coefficients α11 to α16 and α21 to α26 are used to perform the downmix processing based on the formulas (1) and (2) to generate the 2ch acoustic signal S13 (L0, R0).

[Formula 1]
L0 = ± α11 × L ± α12 × R ± α13 × C
± α14 × Ls ± α15 × Rs ± α16 × LFE (1)
R0 = ± α21 × L ± α22 × R ± α23 × C
± α24 × Ls ± α25 × Rs ± α26 × LFE (2)

  In the above formulas (1) and (2), the sign of each downmix coefficient corresponds to the phase difference of each signal. The control circuit 17 may receive the instruction signal S18 indicating various modes input from the input circuit 18, for example, and set the downmix coefficient based on a table or the like previously associated with the mode instruction signal. In the above embodiment, the downmix unit 13 is formed by a multiplier and an adder. However, the present invention is not limited to this mode. For example, various functional units such as a delay unit and a bandpass filter are provided in each channel in front of the adder. By providing, the acoustic signal S13 having a higher quality surround effect can be generated.

  Further, for example, when a 2ch acoustic signal S12 (left front signal L, right front signal R) is input, the downmix unit 13 generates a 2ch signal S13 (L0) based on Equations (1) and (2). , R0). At this time, the downmix unit 13 may simply generate the 2ch signal S13 (L0, R0) by multiplying the acoustic signal S12 (left front signal L, right front signal R) independently by coefficients.

  For example, when the 1ch acoustic signal S12 (center signal C) is input, the downmix unit 13 generates the 2ch signal S13 (L0, R0) based on the formulas (1) and (2). . At this time, the downmix unit 13 may simply generate the left front signal L0 and the right front signal R0 by multiplying the acoustic signal S12 (center signal C) by a predetermined coefficient.

  The downmix unit 13 is not limited to the above-described embodiment. For example, when the audio signal S12 having the number of channels of 6ch or more is input, the downmix unit 13 multiplies the audio signal S12 of each channel by a predetermined downmix coefficient and adds the result to add 2ch. The acoustic signal S13 may be generated.

  The process by the downmix unit 13 corresponds to a virtual surround process, and the acoustic signal S13 (L0, R0) obtained thereby corresponds to a virtual surround signal.

  FIG. 3 is a diagram for explaining the correction unit 30 of the digital audio signal processing apparatus 1 shown in FIG. FIG. 3A is a diagram for explaining the correction unit 30 provided at the output stage of the decoder unit 12, and FIG. 3B is provided between the decoder unit 12 and the downmix unit 13. FIG. 3C is a diagram for explaining the correction unit 30, and FIG. 3C is a diagram for explaining the correction unit 30 provided in the input stage of the downmix unit 13.

  The correcting unit 30 attenuates the level of the acoustic signal S12 of each channel generated by the decoder unit 12 with a specified attenuation rate β and outputs the attenuated signal to the downmix unit 13. The correction unit 30 may be provided, for example, at the output stage of the decoder unit 12 as shown in FIGS. 1 and 3A, or between the decoder unit 12 and the downmix unit 13 as shown in FIG. It may be provided in between, or may be provided in the input stage of the downmix unit 13 as shown in FIG. Further, as a function substantially the same as that of the correction unit 30, the downmix coefficient of the downmix unit 13 may be multiplied by the attenuation rate β. In this embodiment, a correction unit 30 is provided at the output stage of the decoder unit 12 as shown in FIGS.

  The correction unit 30 includes, for example, the same number of multipliers as the number of output channels of the decoder unit 12. Specifically, for example, as illustrated in FIGS. 3A to 3C, the correction unit 30 includes multipliers 301 to 306 corresponding to the acoustic signals S12 of the respective channels. The devices 301 to 306 are set with attenuation rates (attenuation coefficients) β1 to β6. The attenuation rates β1 to β6 are collectively referred to as the attenuation rate β.

  The attenuation rate β is controlled based on the control signal CTL1 input from the control circuit 17, and the correction unit 30 performs a process of attenuating the level of the acoustic signal S12 of each channel with the set attenuation rate β. For example, the control circuit 17 controls the attenuation rate β so that the level of the signal S12 of each channel does not exceed a predetermined maximum level in order to prevent the signal S12 from clipping beyond the predetermined maximum level. To do. At this time, the control circuit 17 may control the attenuation rate β based on the channel number data SD1. Further, the control circuit 17 determines the attenuation factor so that the level of the signal S1 that is finally output does not exceed a predetermined maximum level based on the amplification factors of the high frequency emphasizing unit 14 and the low frequency emphasizing unit 15. β may be controlled.

  As shown in FIG. 1, the emphasizing unit 40 sets the low frequency and / or high frequency level of the acoustic signal S13 generated by the downmix unit 13 (acoustic signal conversion unit) to a specified amplification factor γ. To generate an acoustic signal S1 (L02, R02), and output the acoustic signal S1 (L02, R02) to the DAC 16 and the communication unit 19. The amplification factor γ of the enhancement unit 40 is controlled by a control signal CTL40 (CTL14, CTL15) from the control circuit 17.

  The enhancement unit 40 includes the high frequency enhancement unit 14 and the low frequency enhancement unit 15 as described above. In the emphasis unit 40 according to the present embodiment, the low-frequency emphasis unit 15 is provided after the high-frequency emphasis unit 14, but conversely, the high-frequency emphasis unit 14 may be provided after the low-frequency emphasis unit 15. The functions of the region enhancement unit 14 and the low region enhancement unit 15 may be processed in parallel.

  FIG. 4 is a diagram for explaining a specific example of the high frequency emphasizing unit 14 and the low frequency emphasizing unit 15 of the emphasizing unit 40 shown in FIG.

  The high frequency emphasizing unit 14 amplifies the high frequency level of the input acoustic signal S13 (L0, R0) by the amplification factor γ14 set by the control signal CTL14 by the control circuit 17, and thereby the acoustic signals L01, R01. The sound signals L01 and R01 are generated and output to the low frequency emphasizing unit 15.

  Specifically, the high frequency emphasis unit 14 includes a high pass filter (141L, 141R), a multiplier (142L, 142R), and an adder (143L, 143R) as shown in FIG. 4, for example. The high frequency emphasizing unit 14 having the above configuration extracts a high frequency component from the input left channel acoustic signal L0 by a high pass filter 141L, amplifies the high frequency component by a multiplier 142L at an amplification factor γ14, The amplified high frequency component is added to the original left channel acoustic signal L0 by the adder 143L to generate the acoustic signal L01. Similarly, the high-frequency emphasizing unit 14 having the above configuration extracts a high-frequency component from the input right-channel acoustic signal R0 by a high-pass filter 141R, and amplifies the high-frequency component by an amplification factor γ14 by a multiplier 142R. Then, the amplified high frequency component is added to the original left channel acoustic signal R0 by the adder 143R to generate the acoustic signal R01.

  The low-frequency emphasizing unit 15 amplifies the low-frequency level of the input acoustic signals L01 and R01 by the amplification factor γ15 set by the control signal CTL15 by the control circuit 17, and the acoustic signal S1 (L02, R02). It generates and outputs the acoustic signal S1 (L02, R02) to the DAC 16 and the communication unit 19.

  Specifically, as shown in FIG. 4, for example, the low-frequency emphasizing unit 15 includes a low-pass filter (151L, 151R), a multiplier (152L, 152R), and an adder (153L, 153R). The low frequency emphasis unit 15 having the above configuration extracts a low frequency from the input left channel acoustic signal L01 by the low-pass filter 151L, amplifies the low frequency component by the multiplier 152L at the amplification factor γ15, and the amplification The added low frequency component is added to the original left channel acoustic signal L01 by the adder 153L to generate the acoustic signal L02. Similarly, the low-frequency emphasis unit 15 having the above configuration extracts a low-frequency component from the input right-channel acoustic signal R01 by the low-pass filter 151R, and amplifies the low-frequency component to the multiplier 152R with an amplification factor γ15. Then, the amplified low frequency component is added to the original left channel acoustic signal R01 by the adder 153R to generate the acoustic signal R02.

  FIG. 5 is a diagram for explaining the operation of the enhancement unit 40 of the digital audio signal processing apparatus 1 shown in FIG. The horizontal axis indicates the frequency, and the vertical axis indicates the amplification factor. For example, as illustrated in FIG. 5, the enhancement unit 40 according to the present embodiment amplifies the low frequency level of the signal S13 with the amplification factor γ13 and amplifies the high frequency level of the signal S13 with the amplification factor γ14. Thus, the low frequency and high frequency of the signal S13 are emphasized. For example, the high frequency emphasis unit 14 emphasizes a frequency band of about 1.5 kHz to 96 kHz, and the low frequency emphasis unit 15 emphasizes a frequency band of about 500 Hz or less. The frequency band emphasized by the high frequency emphasizing unit 14 and the low frequency emphasizing unit 15 is appropriately set according to the external environment.

  The digital / analog converter (DAC) 16 converts the digital signal S1 (L02, R02) output from the emphasizing unit 40 into analog signals SAL, SAR and outputs them.

  The communication unit 19 performs data communication with the reception-side device 200 by, for example, a wired method or a wireless method, and outputs a signal S1 to the reception-side device 200.

  The input circuit 18 outputs an instruction signal S18 indicating various modes such as a normal mode and a night mode to the control circuit 17, for example. Examples of the input circuit 18 include a switch, a button, a remote controller, an operation input device such as a mouse and a keyboard, a light receiving sensor such as a photodiode, and a clock (timer circuit). In night mode, for example, when the volume level of the playback sound is set to a relatively low level at night or in a quiet environment, the sound reverberation and clarity are enhanced by emphasizing the high and low frequencies of the sound signal. This is a mode for improving (loudness effect).

  The control circuit 17 comprehensively controls the entire apparatus to realize the functions according to the present invention. For example, the control circuit 17 may realize the function according to the present invention by executing a program stored in a memory (not shown).

  For example, based on the instruction signal S18 input from the input circuit 18, the control circuit 17 functions the correction unit 30 and the enhancement unit 40 when the instruction signal S18 indicating the first mode (night mode) is input, for example. When the instruction signal S18 indicating the second mode (normal mode) different from the first mode is input, the processing by the correction unit 30 and the enhancement unit 40 is suppressed.

  For example, in a general processing apparatus as a comparative example, the digital audio data exceeds the maximum level of 0 dB in order to simply emphasize high and low frequencies with respect to the digital audio data, and the sound is clipped and distorted. Occurs.

  On the other hand, in the control circuit 17 according to the present invention, based on the amplification factor γ of the enhancement unit 40, the level of the acoustic signal S12 of each channel is attenuated by the attenuation factor β by the correction unit 30 and controlled. Specifically, the control circuit 17 controls the correction unit 30 and the enhancement unit 40 to set the levels of the acoustic signals S13 and S1 to a specified level (for example, the maximum level) or less.

  At this time, the control circuit 17 determines the attenuation rate β (β1 to β6) of the correction unit 30 and the amplification of the enhancement unit 40 based on the number of channels of the input source signal and the number of channels of the acoustic signal S12 output from the decoder unit 12. The rate γ (γ14, γ15) is controlled. Specifically, the control circuit 17 controls the attenuation rate β of the correction unit 30 and the amplification rate γ of the enhancement unit 40 based on the channel number data SD1 input from the decoder unit 12.

  As described above, the control circuit 17 controls the level of the acoustic signal S12 of each channel in accordance with the number of channels, so that stereo downmix can be performed while maintaining a good S / N ratio. In addition, sound distortion can be reduced. Further, even when high-frequency emphasis is performed, distortion can be prevented, so that high-quality sound can be reproduced.

  FIG. 6 is a flowchart for explaining an embodiment of the operation of the digital audio signal processing apparatus 1 shown in FIG. The operation of the digital audio signal processing apparatus 1 configured as described above will be described with reference to the drawings. In the present embodiment, a case will be described in which input source signals (digital audio signals) S0 having the number of channels such as 5.1ch (specifically 6ch), 2ch, or 1ch are input to the digital audio signal processing apparatus 1. .

  First, based on the instruction signal S18, the control circuit 17 determines whether or not it is the night mode (ST1), and when it is determined that it is not the night mode (normal mode), the processing of the correction unit 30 and the enhancement unit 40 is suppressed. Then, normal processing is performed (ST2).

  Processing in the normal mode will be described. First, the DIR 11 extracts and demodulates the clock signal (CK) and data (DATA) from the input source signal (digital audio signal) S 0, and outputs the demodulated clock signal (CK) and data DT to the decoder unit 12. . Next, the decoder unit 12 generates an acoustic signal S12 from the data DT input from the DIR 11 according to the clock signal (CK), and outputs the acoustic signal S12 to the downmix unit 13. The downmix unit 13 generates a 2ch acoustic signal S13 (L0, R0) by synthesizing the acoustic signal S12 based on a preset downmix coefficient. The digital acoustic signal S13 is converted into analog signals SAL and SAR by the DAC 16, amplified by the amplifier 3 (3L, 3R), and produced by the speaker 4 (4L, 4R). The acoustic signal S13 is output from the communication unit 19 to the receiving device 200 as necessary, and is generated by the speaker 204 of the receiving device 200.

  On the other hand, if the control circuit 17 determines in step ST1 that the mode is the night mode based on the instruction signal S18 from the input circuit 18, the control unit 17 causes the correction unit 30 and the enhancement unit 40 to function. Next, processing in the night mode will be described. First, the DIR 11 extracts and demodulates the clock signal (CK) and data (DATA) from the input source signal (digital audio signal) S 0, and outputs the demodulated clock signal (CK) and data DT to the decoder unit 12. . Next, the decoder unit 12 generates an acoustic signal S12 having the number of channels corresponding to the number of channels of the input source signal in accordance with the clock signal (CK) from the data DT input from the DIR 11 (ST3). The decoder unit 12 outputs channel number data SD1 indicating the number of channels of the acoustic signal S12 to the control circuit 17.

  The control circuit 17 determines the number of channels of the acoustic signal S12 based on the input channel number data SD1 (ST4). The control circuit 17 determines, for example, the amplification factor γ (γ14, γ15) of the enhancement unit 40 based on the determination result (ST5), and sends the control signals CTL14, 15 for setting the amplification factor γ to the high frequency enhancement unit 14, Output to the low frequency emphasis unit 15. The control circuit 17 also determines the attenuation rate β (β1 to β1) of the correction unit 30 so that the levels of the acoustic signals S13 and S1 are, for example, equal to or lower than a prescribed level (for example, the maximum level) based on the amplification factor γ and the channel number data SD1. β6) is determined (ST6), and a control signal CTL1 for setting the attenuation rate β is output to the correction unit 30.

  Then, the correction unit 30 performs the attenuation process on the acoustic signal S12 of each channel at the attenuation rate β set by the control circuit 17, and outputs the result to the downmix unit 13 (ST7). The downmix unit 13 generates the 2ch acoustic signal S13 (L0, R0) by synthesizing the acoustic signal S12 based on the preset downmix coefficients α11 to α16, α21 to α26 (ST8). . The high frequency emphasizing unit 14 and the low frequency emphasizing unit 15 of the emphasizing unit 40 amplify the low frequency and high frequency levels of the acoustic signal S13 with the amplification factor γ (γ14, γ15) set by the control circuit 17. The acoustic signal S1 (L02, R02) in which the high frequency and the low frequency are emphasized is generated, and the acoustic signal S1 (L02, R02) is output to the DAC 16 and the communication unit 19 (ST9). The digital acoustic signal S13 is converted into analog signals SAL and SAR by the DAC 16, amplified by the amplifier 3 (3L, 3R), and produced by the speaker 4 (4L, 4R). The acoustic signal S13 is output from the communication unit 19 to the receiving device 200 as necessary, and is generated by the speaker 204 of the receiving device 200.

  Specifically, for example, when a 5.1ch source signal S0 is input, the control circuit 17 sets the attenuation rate (attenuation coefficient) (β1, β2, β3, β4, β5, β6) of each channel to ( 0.25, 0.25, 0.25, 0.25, 0.25, 0.25) or less. For example, when a 2-channel source signal S0 is input, the control circuit 17 sets the attenuation rate (attenuation coefficient) (β1, β2, β3, β4, β5, β6) of each channel to (1, 1, 0, 0). , 0, 0) or less. For example, when the 1ch source signal S0 is input, the control circuit 17 sets the attenuation rate (attenuation coefficient) (β1, β2, β3, β4, β5, β6) of each channel to (0, 0, 1, 0). , 0, 0) or less. The attenuation rate β is not limited to the above form.

  For example, when the 5.1 channel source signal S0 is input, for example, the control circuit 17 sets the attenuation rate β6 of the LFE channel to be smaller than the attenuation rates β1, β2, β3, β4, and β5 of the other channels. As a result, it is possible to generate the acoustic signal S1 in which the level of the frequency component that is lower than the frequency component emphasized by the low frequency emphasizing unit 15 is emphasized. The effect that a quality reproduction sound can be obtained is improved.

  As described above, the digital audio signal processing device 1 that generates the acoustic signal S1 having the number of channels defined based on the input source signal S0 according to the present embodiment corresponds to the number of channels of the input source signal S0. Since the processing unit 10 for controlling the level of the acoustic signal of each channel and emphasizing the low and / or high level of the acoustic signal is included, the input source signal S0 with various numbers of channels is input. Even if it exists, the distortion of the signal by clipping can be prevented and high quality acoustic signals S1, SAL, and SAR can be generated. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  Specifically, the digital audio signal processing apparatus 1 generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal S0 based on the input source signal S0, and channel data SD1 indicating the number of channels. Is corrected by the correction unit 30, and the correction unit 30 that attenuates the level of the first acoustic signal S12 generated by the multi-channel decoder unit 12 with the specified attenuation factor β. A stereo downmix unit 13 (acoustic signal conversion unit) 13 that generates a second acoustic signal S13 having a predetermined number of channels based on the first acoustic signal S12, and a stereo downmix unit (acoustic signal conversion unit) The level of both or one of the low and high frequencies of the second acoustic signal S13 generated by A control circuit (control unit) 17 that controls the attenuation rate β of the correction unit 30 and the amplification rate γ of the enhancement unit 40 based on the enhancement unit 40 that amplifies at a rate and the channel number data SD1 of the first acoustic signal S12. Therefore, even when input source signals S0 having various numbers of channels are input, distortion of signals due to clipping can be prevented and high-quality acoustic signals S1, SAL, and SAR can be generated. In addition, it is possible to improve the reverberation and clarity of the reproduced sound during small volume reproduction.

  Further, the control circuit 17 controls the attenuation rate β of the correction unit 30 based on the amplification factor γ of the emphasizing unit 40 so that the levels of the acoustic signal S13 and the acoustic signal S1 are not more than a specified level. Signal distortion can be prevented and high-quality sound can be reproduced.

  Further, for example, when the instruction signal S18 indicating the first mode (night mode) is input from the input circuit 18, the control circuit 17 causes the correction unit 30 and the enhancement unit 40 to function and is different from the first mode. When the instruction signal S18 indicating the second mode (normal mode) is input, the processing by the correcting unit 30 and the emphasizing unit 40 is suppressed. Therefore, the digital audio signal processing apparatus 1 performs the normal processing in the normal mode, For example, when listening to content such as a movie at night, etc., the amplification levels of the amplifiers 3, 3L, 3R, 203 (203L, 203R) are set relatively low in consideration of the surroundings, and the volume levels of the reproduced sound are compared. If the instruction signal S18 indicating the night mode is input from the input circuit 18 by a user operation, the correction unit 30 and the enhancement unit 40 function. Without causing clipping, it is possible to emphasize the high-frequency component and low-frequency component of the reproduced sound, it is possible to reliably improve the clarity of the sound of the sound and speech.

  In addition, the digital audio signal processing apparatus 1 according to the present embodiment generates 2ch acoustic signals S1, SAL, and SAR based on the input source signal S0, so that, for example, 2ch acoustic signals are reproduced by left and right speakers, for example, headphones. By doing so, the listener can hear the realistic 3D sound similar to the surround system.

[Second Embodiment]
FIG. 7 is a functional block diagram for explaining an audio system 100a employing the digital audio signal processing device 1a according to the second embodiment of the present invention. The description of the same configuration, function, operation, etc. as in the first embodiment is omitted.

  When the digital audio signal processing device 1a according to the present embodiment generates an acoustic signal having a specified number of channels based on the input source signal S0, the input source signal S0 has a format that can control the dynamic range. The dynamic range is controlled to emphasize both or one of the low and high frequency levels of the acoustic signal with the first amplification factor, and when the input source signal S0 is in any other format, the acoustic signal is low. There is a processing unit 10a that emphasizes both or one of the high frequency range and the high frequency level with a second gain larger than the first gain.

  Specifically, as shown in FIG. 7, the digital audio signal processing apparatus 1a includes a digital audio interface receiver (DIR) 11, a multichannel decoder unit (decoder unit: MCD) 12a, a correction unit 30a, and a stereo downmix unit. (Downmix section: SDMP) 13, digital audio data high frequency enhancement section (high frequency enhancement section: TB (Treble boost)) 14, digital audio data low frequency enhancement section (low frequency enhancement section: BB (Bass boost)) 15 A digital / analog converter (DAC) 16 and a control circuit (MP: microprocessor) 17a.

  For example, the processing unit 10 a includes a DIR 11, a decoder unit 12 a, a correction unit 30 a, a downmix unit 13, an emphasizing unit 40 (high frequency emphasizing unit 14, low frequency emphasizing unit 15), and a control circuit 17 a. Corresponds to an embodiment of the processing unit according to the present invention.

  Based on the input source signal (digital audio data) S0, the decoder unit 12a generates an acoustic signal S12 having the number of channels corresponding to the number of channels of the input source signal S0. In addition, the decoder unit 12a outputs audio format information SD1a indicating the audio format of the input source signal S0 to the control circuit 17a. At this time, the decoder unit 12a may output, as the audio format information SD1a, for example, audio format information SD1a included in the source signal S10 or data indicating the audio format included in the data DT by the DIR11.

  In the source signal (digital audio data) S0, an acoustic signal is encoded in various formats defined in advance. The audio format information SD1a indicating the audio format can include various formats such as Dolby digital audio format (AC3: Audio Code number 3), PCM (Pulse Code Modulation), AAC (Advanced Audio Coding), and the like.

  Examples of audio formats that can control the dynamic range according to the present invention include various formats such as Dolby Digital Audio Format (AC3). Examples of audio formats whose dynamic range cannot be controlled include various formats such as PCM and AAC.

  The correction unit 30a controls the dynamic range of the acoustic signal S12 output from the decoder unit 12a under the control of the control circuit 17a.

  For example, when the instruction signal S18 indicating the night mode is input from the input circuit 18, the control circuit 17a determines the dynamic range of the acoustic signal S12 generated by the decoder unit 12a based on the format information SD1a of the input source signal S0. When it is determined that control is possible, the dynamic range of the acoustic signal S12 is controlled by the correction unit 30a and the enhancement unit 40 is amplified with the first amplification factor γ1, and in other cases, the correction unit The control by 30a is suppressed, and the emphasis unit 40 is made to amplify at a second gain γ2 that is larger than the first gain γ1.

  As described above, compared with the amplification factor γ1 of the enhancement unit 40 when the input source signal S0 is an audio format capable of dynamic range control, the enhancement unit when the input source signal S0 is an audio format where dynamic range control is not possible. Since the amplification factor γ2 of 40 is large, the difference in the effect of the night mode due to the difference in audio format can be reduced.

  FIG. 8 is a diagram for explaining the operation of the correction unit 30a of the digital audio signal processing apparatus 1a shown in FIG. FIG. 8A is a diagram for explaining the operation when the dynamic range is not compressed, and FIG. 8B is a diagram for explaining the operation when the dynamic range is compressed.

  Specifically, when the control circuit 17a determines that the dynamic range of the acoustic signal S12 generated by the decoder unit 12a can be controlled based on the format information SD1a of the input source signal S0, FIG. As shown in (B), the dynamic range of the acoustic signal S12 is controlled by the correction unit 30a. At this time, as shown in FIG. 8B, the output level by the correction unit 30a responds in an S-shaped curve to the input level (level before conversion) of the acoustic signal S12, and the output level range (Lu˜ The dynamic range is compressed to Ld).

  On the other hand, when the control circuit 17a determines that the dynamic range of the acoustic signal S12 generated by the decoder unit 12a cannot be controlled based on the format information SD1a of the input source signal S0, FIG. As shown in FIG. 3, the dynamic range of the acoustic signal S12 is controlled by the correction unit 30a. At this time, as shown in FIG. 8A, the output level by the correction unit 30a responds substantially linearly to the input level (level before conversion) of the acoustic signal S12, and the output level range (La to Lb). Is set to

  Note that the output upper limit level Lu described above is set smaller than the maximum level La, and the output lower limit level Ld is set larger than the minimum level Lb. That is, the ratio between the output upper limit level Lu and the output lower limit level Ld is smaller than the ratio between the maximum level La and the minimum level Lb.

  By compressing the dynamic range, the level of a relatively loud sound is reduced, and the level of a relatively low sound is increased. Even in the case of, it is possible to eliminate difficulty in hearing such as lines.

  FIG. 9 is a flowchart for explaining the operation of the correction unit 30a of the digital audio signal processing apparatus 1a shown in FIG. The operation of the digital audio signal processing apparatus 1a having the above configuration will be described with reference to FIG. Description of operations similar to those of the first embodiment is omitted.

  First, based on the instruction signal S18, the control circuit 17a determines whether or not it is the night mode (ST1), and if it is determined that it is not the night mode (normal mode), the processing of the correction unit 30a and the enhancement unit 40 is suppressed. Then, normal processing is performed (ST2).

  On the other hand, when the control circuit 17 determines that the night mode is set based on the instruction signal S18 from the input circuit 18 in step ST1, the control unit 17 causes the correction unit 30a and the enhancement unit 40 to function. Next, processing in the night mode according to the present embodiment will be described. First, the DIR 11 extracts and demodulates the clock signal (CK) and data (DATA) from the input source signal (digital audio signal) S0, and outputs the demodulated clock signal (CK) and data DT to the decoder unit 12a. . Next, the decoder unit 12a generates the acoustic signal S12 having the number of channels corresponding to the number of channels of the input source signal from the data DT input from the DIR 11 according to the clock signal (CK). The decoder unit 12 outputs the format information SD1a of the source signal S0 to the control circuit 17a (ST13).

  Next, the control circuit 17a determines whether or not the dynamic range of the acoustic signal S12 generated by the decoder unit 12a can be controlled based on the format information SD1a of the input source signal S0 (ST14). When it is determined that control is possible, the dynamic range of the acoustic signal S12 is controlled by the correction unit 30a (ST15) as shown in FIG. 8B, and the first amplification factor γ1 is set to the enhancement unit 40. The control signal CTL40 (CTL14, 15) to be set is output and amplified by the high frequency emphasizing unit 14 and the low frequency emphasizing unit 15 of the emphasizing unit 40 with the amplification factor γ1 (ST16).

  On the other hand, when the control circuit 17a determines in step ST14 that the dynamic range of the acoustic signal S12 generated by the decoder unit 12a cannot be controlled based on the format information SD1a of the input source signal S0. A control signal CTL40 (CTL14, 15) that suppresses the control by the correction unit 30a and causes the enhancement unit 40 to set a second amplification factor γ2 that is larger than the first amplification factor γ1 is output. The unit 14 and the low frequency emphasizing unit 15 are amplified with an amplification factor γ2 (ST17).

  As described above, the digital audio signal processing apparatus 1a according to the present embodiment controls the dynamic range by the correcting unit 30a and controls the sound when the input source signal S0 has a format in which the dynamic range can be controlled in the night mode. If the enhancement unit 40 emphasizes the low frequency and / or high frequency level of the signal with the first amplification factor γ1, and the input source signal S0 is in any other format, the low frequency and high frequency of the acoustic signal Is provided with the control circuit 17a that causes the emphasis unit 40 to emphasize both or one of the levels at a second amplification factor γ2 larger than the first amplification factor γ1, that is, when dynamic compression is performed, the maximum volume and the minimum volume. Even if the playback is performed at a low volume, such as at night, it is not too loud for the user and the sound is extremely low. There is an effect that it is not. Furthermore, in the present embodiment, the clarity of the sound is further improved by emphasizing the high sound range and the low sound range.

  Further, the amplification factor γ1 of the enhancement unit 40 when the input source signal S0 is an audio format in which the dynamic range control is not possible, compared with the amplification factor γ1 of the enhancement unit 40 when the input source signal S0 is an audio format in which the dynamic range control is possible. Since γ2 is large, the difference in enhancement effect due to the difference in audio format can be reduced.

  In other words, when the input source signal S0 is an audio format in which the dynamic range control is possible and an audio format in which the dynamic range control is not possible, when the high frequency range and the low frequency range are simply emphasized with the same amplification factor γ, Since the enhancement process is performed twice, there is a difference in the enhancement effect due to the difference in the audio format. However, in the present invention, there is no difference in the enhancement effect between the case where the dynamic range compression is performed and the case where the dynamic range compression is not performed according to the audio format. In addition, by controlling the amplification factor γ by the enhancement unit 40, the difference in enhancement effect due to the difference in audio format can be reduced.

  The present invention is not limited to the embodiment described above. You may combine embodiment and the specific example which were mentioned above.

1 is a functional block diagram for explaining an audio system 100 employing a digital audio signal processing device 1 according to a first embodiment of the present invention. It is a circuit diagram for demonstrating the downmix part 13 which concerns on one Embodiment of the digital audio signal processing apparatus 1 shown in FIG. It is a figure for demonstrating the correction | amendment part 30 of the digital audio signal processing apparatus 1 shown in FIG. (A) is a figure for demonstrating the correction | amendment part 30 provided in the output stage of the decoder part 12, (B) shows the correction | amendment part 30 provided between the decoder part 12 and the downmix part 13. FIG. It is a figure for demonstrating, (C) is a figure for demonstrating the correction | amendment part 30 provided in the input stage of the downmix part 13. FIG. It is a figure for demonstrating one specific example of the high-frequency emphasis part 14 and the low-frequency emphasis part 15 of the emphasis part 40 shown in FIG. It is a figure for demonstrating operation | movement of the emphasis part 40 of the digital audio signal processing apparatus 1 shown in FIG. 3 is a flowchart for explaining an embodiment of the operation of the digital audio signal processing apparatus 1 shown in FIG. 1. It is a functional block diagram for demonstrating the audio system 100a which employ | adopted the digital audio signal processing apparatus 1a which concerns on 2nd Embodiment of this invention. It is a figure for demonstrating operation | movement of the correction | amendment part 30a of the digital audio signal processing apparatus 1a shown in FIG. (A) is a figure for demonstrating the operation | movement at the time of dynamic range non-compression, (B) is a figure for demonstrating the operation | movement at the time of dynamic range compression. It is a flowchart for demonstrating operation | movement of the correction | amendment part 30a of the digital audio signal processing apparatus 1a shown in FIG.

Explanation of symbols

1 Digital Audio Signal Processing Device 2 Audio Device (Digital Audio Device)
10 Processing Unit 11 Digital Audio Interface Receiver (DIR)
12, 12a Multi-channel decoder unit (decoder unit: MCD)
13 Stereo downmix section (downmix section: SDMP) (acoustic signal conversion section)
14 Digital audio data high frequency enhancement part (Treble boost: TB)
15 Digital audio data low-frequency emphasis part (Bass boost: BB)
16 Digital / analog converter (DAC)
17, 17a Control circuit (MP: microprocessor)
18 Input Circuit 19 Communication Unit 30, 30a Correction Unit 40 Emphasis Unit (High Frequency Enhancement Unit 14, Low Frequency Enhancement Unit 15)
100,100 Audio system 200 Receiver device

Claims (8)

A digital audio signal processing device that generates acoustic signals of a specified number of channels based on an input source signal,
A digital audio comprising: a processing unit that controls the level of an acoustic signal of each channel in accordance with the number of channels of the input source signal and emphasizes both or one of the low and high frequencies of the acoustic signal. Signal processing device. A digital audio signal processing device that generates acoustic signals of a specified number of channels based on an input source signal,
When the input source signal has a format in which the dynamic range can be controlled, the dynamic range is controlled to emphasize both or one of the low and high frequencies of the acoustic signal with a first amplification factor. When the source signal has a format other than the above, it has a processing unit that emphasizes both or one of the low frequency and high frequency levels of the acoustic signal with a second amplification factor larger than the first amplification factor. A digital audio signal processing apparatus. Based on the input source signal, a multi-channel decoder unit that generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal, and a predetermined number of channels based on the first acoustic signal A digital audio signal processing apparatus having an acoustic signal conversion unit for generating the second acoustic signal,
A correction unit that attenuates the level of the first acoustic signal generated by the multi-channel decoder unit at a specified attenuation rate;
An emphasis unit that amplifies the low frequency and / or high frequency level of the second acoustic signal generated by the acoustic signal conversion unit at a specified amplification factor;
A digital audio signal processing apparatus comprising: a control unit that controls an attenuation rate of the correction unit and an amplification rate of the enhancement unit based on the number of channels of the first acoustic signal.   The digital audio signal processing apparatus according to claim 3, wherein the control unit controls an attenuation rate of the correction unit based on an amplification factor of the enhancement unit. Based on the input source signal, a multi-channel decoder unit that generates a first acoustic signal having the number of channels corresponding to the number of channels of the input source signal, and a predetermined number of channels based on the first acoustic signal A digital audio signal processing apparatus having an acoustic signal conversion unit for generating the second acoustic signal,
A correction unit for controlling a dynamic range of the first acoustic signal;
An emphasis unit that amplifies the low frequency and / or high frequency level of the second acoustic signal generated by the acoustic signal conversion unit at a specified amplification factor;
When it is determined that the dynamic range of the first acoustic signal can be controlled based on the format information of the input source signal, the correction unit controls the dynamic range of the first acoustic signal. In addition, the enhancement unit amplifies at a first amplification factor, otherwise, the control by the correction unit is suppressed and the enhancement unit amplifies at a second amplification factor larger than the first amplification factor. A digital audio signal processing device.   6. The digital according to claim 3, wherein the control unit controls the correction unit and the enhancement unit so that the level of the second acoustic signal is equal to or lower than a specified level. Audio signal processing device.   When the instruction signal indicating the first mode is input, the control unit causes the correction unit and the enhancement unit to function, and when the instruction signal indicating the second mode different from the first mode is input The digital audio signal processing apparatus according to claim 3, wherein processing by the correction unit and the enhancement unit is suppressed.   8. The digital audio signal processing apparatus according to claim 1, wherein a two-channel acoustic signal is generated based on the input source signal.

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