JP2007079306A - Audio signal processing apparatus and audio signal processing method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an audio signal processing apparatus capable of automatically setting and performing optimal sound quality improvement processing corresponding to the bit rate of a compressed audio file. <P>SOLUTION: The audio signal processing apparatus comprises an automatic sound quality improvement processing means for setting and performing sound quality improvement processing by selecting parameter values of parameters P1-P5 out of a plurality of predetermined stages in accordance with the bit rate of a compressed audio file F as one addition process of higher harmonic wave component addition processing for adding a lost higher harmonic wave component to an audio signal resulting from decoding the compressed audio file F compressing information based on parameter values of a higher harmonic wave component generation bandwidth parameter P1 and a higher harmonic wave component addition level parameter P2, peak component addition processing for adding a dull peak value based on a parameter value of a peak component correction level parameter P3, and attack component addition processing for adding a lost attack component based on parameter values of an attach component generation bandwidth parameter P4 and an attack component addition level parameter P5. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention demodulates a compressed audio file recorded on a digital recording medium such as a hard disk (HD) or a mini disk (MD) or a compressed audio file transmitted via a communication line and recorded in a memory device. It belongs to the technical field of audio signal processing in an audio playback device such as a portable audio player for playback.

MP3 (MPeg audio layer 3), WMA (Windows (registered trademark) Media Audio), AAC (Advanced Audio Coding), ATRAC (Adaptive TRansform Acoustic Coding)
A compressed audio file obtained by compressing information of a digital audio signal by a compression technique such as the above generally deteriorates in sound quality when the bit rate (unit: bps) is lowered to increase the compression rate. That is, the harmonic (high frequency) component is largely lost, the peak component is dull, and the low frequency and high frequency attack components are also dull.

  However, a digital compressed audio file recorded on a hard disk (HD) or mini disk (MD) or a compressed audio file transmitted via a communication line such as the Internet and recorded in a memory device such as a flash ROM. In an audio playback device that demodulates (expands), amplifies, converts the signal to analog, and outputs the sound from an earphone or speaker, the signal is added by the equivalent of the harmonic component of the original digital audio signal lost due to compression or lost due to compression. Some sound quality enhancement processing such as adding an attack component of a broken attack sound is performed.

  For example, in a conventional portable audio player, regardless of the bit rate of the compressed audio file, the sound quality enhancement process is performed with the same degree of effect or the parameter indicating the degree of the effect of the sound quality improvement process is set. The parameter value is determined manually by the user (selection switch operation for the degree of the effect of high-quality sound processing in several steps such as strong, medium, and weak).

  By the way, as a known document focusing on the relationship between the bit rate of a compressed audio file and sound quality improvement, for example, the following [Patent Document 1] is adjacent on the frequency axis in order to perform highly efficient compression and expansion. As a configuration in which a plurality of scale factors sf are referenced, a scale factor representative value SF is obtained based on the scale factor sf, and quantization is performed with the bit allocation amount obtained based on the scale factor representative value SF. With the adaptive bit allocation method utilizing the masking effect and / or interband masking effect, good sound quality can be obtained at the same bit rate, and sound quality degradation can be prevented at a low bit rate. Techniques for recording, reproducing and transmitting compressed digital audio data have been proposed.

Japanese Patent Laid-Open No. 06-318875

  When a high-bit-rate compressed audio file and a low-bit-rate compressed audio file coexist in data recorded on a recording medium such as HD, MD, or flash ROM, or in the middle of a single compressed audio file When the rate changes, it is very troublesome for the user to manually change and set the parameter values of various parameters indicating the degree of high sound quality effect to the optimum state at the beginning of playback. It is difficult to judge whether or not the setting is appropriate.

  In this respect, audio playback devices for compressed audio files, especially portable audio players, place importance on design and operability, and minimize the number of operation buttons and switches, and the effects of various high-quality sound processing. It is desirable that the degree is always automatically set to the optimum state.

  However, the technique of the adaptive bit allocation method in the compressed data recording, reproduction, and transmission described in [Patent Document 1] is based on the compression technique at the time of creating the compressed audio file and is input. There is no useful suggestion about automating the optimal settings for high quality sound processing when reproducing and outputting existing compressed audio files.

  The present invention has been made in view of the above circumstances, and a high quality sound processing apparatus in an audio reproduction apparatus that demodulates (expands) a compressed audio file recorded in a memory device such as an HD, MD, or flash ROM, and reproduces and outputs it. It is an object to provide a sound quality improvement processing method.

In order to solve the above problems, the present invention
(1) A harmonic component generation bandwidth parameter P1 and a harmonic component addition level for adjusting a harmonic component lost by information compression with respect to an audio signal obtained by decoding a compressed audio file F obtained by compressing information of a digital audio signal. A harmonic component addition process for adding a harmonic component newly generated based on each parameter value with the parameter P2, or a parameter value of the peak component correction level parameter P3 to adjust a peak value blunted by information compression Each of the parameter values of the attack component generation bandwidth parameter P4 and the attack component addition level parameter P5 in order to adjust the attack component lost due to the information compression or the peak component addition processing for adding the peak component newly generated based on Attack component addition processing for adding newly generated attack components based on According to the bit rate (low, medium, high) of the compressed audio file detected from the compressed audio file F, the high quality sound processing that is at least one of the three additional processes An audio signal comprising high-quality sound processing means for selecting and setting a parameter value of each parameter P1 to P5 from a plurality of predetermined values (large, medium, small, none) A processing device 12 is provided.
(2) For the audio signal obtained by decoding the compressed audio file F obtained by compressing the information of the digital audio signal, the harmonic component generation bandwidth parameter P1 and the harmonic component addition level are used to adjust the harmonic component lost by the information compression. A harmonic component addition processing unit 21 for adding a harmonic component newly generated based on each parameter value with the parameter P2, or a parameter of a peak component correction level parameter P3 for adjusting a peak value blunted by information compression A peak component addition processing unit 22 for adding a peak component newly generated based on the value, or an attack component generation bandwidth parameter P4 and an attack component addition level parameter P5 for adjusting an attack component lost due to information compression; An attack that adds a newly generated attack component based on each parameter value of In accordance with the bit rate of the compressed audio file detected from the compressed audio file F, and the high quality sound processing unit including at least one additional processing unit of the three additional processing units The parameter values in the parameters P1 to P5 of the additional processing unit included in the high sound quality processing units 21, 22, and 23 are selected from a plurality of predetermined values (large, medium, small, and none). An audio signal processing apparatus 12 is provided, comprising: a high sound quality processing control unit 25 configured to control the high sound quality processing units 21, 22, and 23 to perform high sound quality processing. .
(3) Step S1 for inputting and decoding a compressed audio file F obtained by compressing information of a digital audio signal; Step S2 for extracting a bit rate of the compressed audio file F; and adjusting harmonic components lost due to information compression. In order to achieve this, the harmonic component addition processing for adding the newly generated harmonic component based on the parameter values of the harmonic component generation bandwidth parameter P1 and the harmonic component addition level parameter P2 or the information compression Peak component addition processing for adding a peak component newly generated based on the parameter value of the peak component correction level parameter P3 to adjust the peak value, or attack component for adjusting the attack component lost by information compression Each parameter of the generation bandwidth parameter P4 and the attack component addition level parameter P5 Detected from the compressed audio file F with respect to the sound quality enhancement process which is at least one of the three additional processes of the attack component addition process for adding a newly generated attack component based on the data value According to the bit rate (low, medium, high) of the compressed audio file F, the parameter values of the parameters P1 to P5 are selected from a plurality of predetermined values (large, medium, small, none) and set. There is provided an audio signal processing method comprising: Step S3; and Step S4 for performing sound quality enhancement processing based on the set parameter values of the parameters P1 to P5 of the sound quality enhancement processing.

The audio signal processing device and the audio signal processing method according to the present invention are configured as described above.
(1) Since the sound quality enhancement processing for the compressed audio file compressed at various bit rates is performed by automatically setting the parameter value of the parameter indicating the degree of the effect to an appropriate value corresponding to the bit rate, Therefore, an audio output in which appropriate high-quality sound processing is always performed on a compressed audio file having a bit rate can be obtained without bothering the user to perform a switch operation.
(2) The parameter value of the parameter that determines the degree of the effect of appropriate sound quality enhancement processing is automatically selected according to the bit rate of various compressed audio files, and the harmonic component addition processing, peak component addition processing, or attack component is selected. The high quality sound processing, which is at least one of the three additional processes, is appropriately executed, and a high quality sound signal corresponding to the bit rate of the compressed sound file is output.

  Embodiments of an audio signal processing device and an audio signal processing method according to the present invention will be described with reference to the drawings.

  FIG. 1 is a block diagram centering on an audio signal processing apparatus according to the present invention in an audio reproducing apparatus. FIG. 2 is a block diagram of the sound quality enhancement process of the audio signal processing apparatus according to the present invention in the audio reproduction apparatus. FIG. 3 is a schematic waveform diagram before and after the sound quality enhancement process for explaining the outline of the three sound quality enhancement processes. FIG. 4 is a flowchart of the audio signal processing method according to the present invention.

  An audio playback device 20 shown in FIG. 1 is a portable audio player having an HD as a built-in recording medium, and includes a hard disk driver 1 that drives a hard disk HD in which a compressed audio file F is recorded, and a flash that stores a system program. ROM 2, SDRAM 3 as a temporary storage memory, liquid crystal display (LCD) 4, key input device 5, DA converter 6, reproduced audio signal output terminal jack 7, and compressed audio file F from outside For digital signal processing comprising a USB connector 8 for input, a decoder 11 for MP3 and WMA compressed audio file F, an audio signal processing device 12 according to the present invention, and an acoustic effect processing device 13 for surround processing and equalizer processing. System LSI 10 with 1-chip microcomputer as the core , Other has a block configuration comprising a battery 16 and the constant voltage supply circuit 17. The above block diagram is an example, and the flash ROM 2 and the DA converter 6 may be integrated in the system LSI 10 in order to reduce the size and power consumption.

  The audio signal processing device 12 in the audio playback device 20 generates harmonic components for adjusting the harmonic components lost by the information compression with respect to the audio signals obtained by decoding the compressed audio file F obtained by compressing the digital audio signals. To adjust harmonic component addition processing for adding a newly generated harmonic component based on the parameter values of the bandwidth parameter P1 and the harmonic component addition level parameter P2, or the peak value blunted by information compression Peak component addition processing for adding a newly generated peak component based on the parameter value of the peak component correction level parameter P3, or an attack component generation bandwidth parameter P4 and an attack for adjusting an attack component lost by information compression Based on each parameter value with the component addition level parameter P5 The bit rate of the compressed audio file detected from the compressed audio file F, which is at least one of the three additional processes of the attack component adding process for adding the generated attack component. High quality sound processing performed by selecting and setting parameter values of the respective parameters P1 to P5 according to (low, medium, high) from a plurality of predetermined values (large, medium, small, none) Means.

  That is, when the configuration of the audio signal processing device 12 according to the present invention is represented by processing blocks, as shown in FIG. 2, the compressed audio file F obtained by compressing the digital audio signal is compressed by information compression. Harmonic component addition processing for adding a newly generated harmonic component based on the parameter values of the harmonic component generation bandwidth parameter P1 and the harmonic component addition level parameter P2 in order to adjust the lost harmonic component 21 or a peak component addition processing unit 22 for adding a peak component newly generated based on the parameter value of the peak component correction level parameter P3 in order to adjust a peak value blunted by information compression, or by information compression In order to adjust the lost attack component, the attack component generation bandwidth parameter P4 and the attack component addition level And a sound quality enhancement processing unit which is at least one of the three additional processings of the attack component addition processing unit 23 for adding a newly generated attack component based on each parameter value with the parameter P5. In accordance with the bit rate of the compressed audio file detected from the compressed audio file F, the parameter values of the parameters P1 to P5 of the at least one sound quality enhancement processing unit 21, 22, 23 are determined in advance. ROM, etc.) Control is performed so that the sound quality enhancement processing units 21, 22, 23 perform the sound quality enhancement processing by selecting and setting from a plurality of values (large, medium, small, none) recorded in the ROM 26) And a sound quality enhancement processing control unit 25 mainly composed of a CPU of the microcomputer.

  [Table 1] below shows the effects of three sound quality enhancement processes (harmonic component addition process, peak component addition process, attack component addition process) according to the bit rate of the compressed audio file F detected by the decoder 11. It is the parameter combination table | surface which changed the degree of. ◎ is the parameter value setting with the highest degree of effect, ○ is the parameter value setting with the medium degree of effect, △ is the parameter value setting with the low degree of effect, and × is the high sound quality This is a setting that does not process (no assignment). The low bit rate (high compression rate) in the table corresponds to, for example, 8 to 96 kbps for MP3 and 48 kbps for WMA, and the medium bit rate (medium compression rate) is 112 to 192 kbps for MP3, 64 for WMA, This corresponds to 96 and 128 kbps, and the high bit rate (low compression ratio) is set to correspond to 224 to 320 kbps for MP3 and 160 and 192 kbps for WMA. The degree of the effect of sound quality improvement processing is divided.

本発明の音声信号処理装置１２では、前記デコーダ１１で検出された圧縮音声ファイルのビットレートに応じた高音質化処理の最適設定の自動化を実現するために、［表１］から判るように、圧縮率の高い低ビットレートの場合は、情報圧縮による高域成分、アタック成分の劣化（喪失）の程度が大きいので、擬似的・人工的な所為であるものの、より大きな補充・補正、即ち大きな高音質化処理の効果を与える設定になっており、逆に圧縮率の低い高ビットレートの場合は、情報圧縮による音声劣化の程度が小さいので、敢えて高音質化の処理は抑え目（若しくは処理無し）にして、より自然な音質を生かす設定になっているのである。

In the audio signal processing apparatus 12 of the present invention, as can be seen from [Table 1], in order to realize the automatic setting of the optimum sound quality improvement processing according to the bit rate of the compressed audio file detected by the decoder 11, In the case of a low bit rate with a high compression rate, the degree of deterioration (loss) of the high-frequency component and attack component due to information compression is large. It is set to give the effect of high sound quality processing. Conversely, in the case of a high bit rate with a low compression rate, since the degree of sound deterioration due to information compression is small, the high sound quality processing is intentionally suppressed (or processing) None), and it is set to take advantage of the more natural sound quality.

  [Table 2] below is a combination table of setting parameter values of the harmonic component generation bandwidth parameter P1 and the harmonic component addition level parameter P2 of the harmonic component addition processing which is one of the sound quality enhancement processing.

上記高調波成分付加処理は、図４の（ａ）の波形模式図に示されるように、元のデジタル音声信号の波形Ａ（図では矩形波）を情報圧縮すると、その高調波成分の波形が鈍って圧縮波形Ｂのようになるが、この圧縮波形Ｂのデコードされた或る１点のデータとその前後数点のデコードされたデータの変化を順次比較して所定のアルゴリズムに基づいて鈍化した部分のレベルを持ち上げる（付加する）デジタル信号処理を行い、元の波形Ａに近似する波形Ｃを作る高音質化処理である。この効果の度合いを決めるのが上記高調波成分生成帯域幅パラメータＰ１（対象とする高調波成分の生成帯域幅を決めるものであり、広いほど効果の度合いは大きくなる。）と高調波成分付加レベルパラメータＰ２（どの程度レベルを持ち上げるかを決めるパラメータであり、大きいほど効果の度合いは大きくなる。）である。

In the harmonic component addition processing, as shown in the waveform schematic diagram of FIG. 4A, when the waveform A (rectangular wave in the figure) of the original digital audio signal is information-compressed, the waveform of the harmonic component is changed. Although it becomes dull and becomes a compressed waveform B, the data of one decoded point of the compressed waveform B and the change of the decoded data at several points before and after it are sequentially compared and blunted based on a predetermined algorithm. This is a process for improving the sound quality by performing digital signal processing that raises (adds) the level of a portion to create a waveform C that approximates the original waveform A. The degree of this effect is determined by the harmonic component generation bandwidth parameter P1 (which determines the generation bandwidth of the target harmonic component, and the greater the effect, the greater the effect). Parameter P2 (a parameter that determines how much the level is raised, the greater the effect, the greater the effect).

  Next, [Table 3] is a parameter value setting table for the peak component correction level parameter P3 in the peak component addition process, which is one of the sound quality enhancement processes.

上記ピーク成分付加処理は、図４の（ｂ）のサンプリングデータ波形模式図に示されるように、元のデジタル音声信号のサンプリング波形Ａを情報圧縮すると、そのピーク成分が鈍ってサンプリング波形Ｂのようになるが、このサンプリング波形Ｂのピーク値のデータのレベルを所定のアルゴリズムに基づいて（例えばピーク成分を抽出してその尖鋭度を測り、それに応じてレベルを所定割合で）持ち上げる（付加する）デジタル信号処理を行い、元のピーク値に近似するピーク値のサンプリング波形Ｃを作る高音質化処理である。この効果の度合いを決めるのが上記ピーク成分補正レベルパラメータＰ３（どの程度ピーク値のレベルを持ち上げるかを決めるパラメータであり、大きいほど効果の度合いは大きくなる。）である。

In the peak component addition process, as shown in the sampling data waveform schematic diagram of FIG. 4B, when the sampling waveform A of the original digital audio signal is compressed, the peak component becomes dull and the sampling waveform B However, the level of the peak value data of the sampling waveform B is raised (added) based on a predetermined algorithm (for example, the peak component is extracted and its sharpness is measured, and the level is increased at a predetermined rate accordingly). This is a sound quality enhancement process that performs digital signal processing and creates a sampling waveform C having a peak value that approximates the original peak value. The degree of this effect is determined by the peak component correction level parameter P3 (a parameter that determines how much the level of the peak value is raised, and the degree of effect increases as the value increases).

  Next, [Table 4] is a set combination table of parameter values of the attack component generation bandwidth parameter P4 and the attack component addition level parameter P5 of the attack component addition process which is one of the sound quality enhancement processes.

上記アタック成分付加処理は、図４の（ｃ）の波形模式図に示されるように、元のデジタル音声信号の波形Ａを情報圧縮すると、そのアタック成分の波形が鈍って圧縮波形Ｂのようになるが、この圧縮波形Ｂのアタック成分を抽出し、抽出されたアタック成分のレベルと基準レベルとを所定のアルゴリズムで比較してレベルを持ち上げる（付加する）デジタル信号処理を行い、元の波形Ａに近似する波形Ｃを作る高音質化処理である。この効果の度合いを決めるのが上記アタック成分生成帯域幅パラメータＰ４（対象とするアタック成分の生成帯域幅を決めるものであり、広いほど効果の度合いは大きくなる。例えば、◎では〜３００Ｈｚと１ｋＨｚ〜とし、○では〜２００Ｈｚと１．５ｋＨｚ〜とする。）とアタック成分付加レベルパラメータＰ５（どの程度アタック成分のレベルを持ち上げるかを決めるパラメータであり、大きいほど効果の度合いは大きくなる。）である。

In the attack component adding process, as shown in the waveform schematic diagram of FIG. 4C, when the waveform A of the original digital audio signal is information-compressed, the waveform of the attack component becomes dull and the compressed waveform B is obtained. However, the attack component of this compressed waveform B is extracted, the level of the extracted attack component and the reference level are compared with a predetermined algorithm, and digital signal processing for raising (adding) the level is performed, and the original waveform A Is a sound quality enhancement process for creating a waveform C approximating to. The degree of this effect is determined by the attack component generation bandwidth parameter P4 (determining the generation bandwidth of the target attack component. The wider the effect, the greater the effect. , And ◯ is about 200 Hz and 1.5 kHz.) And an attack component addition level parameter P5 (a parameter that determines how much the level of the attack component is raised, and the greater the value, the greater the effect). .

  In the audio signal processing apparatus 12 according to the present invention, at least one of the above three sound quality enhancement processes may be performed. Of course, a plurality of sound quality enhancement processes are combined as shown in [Table 1]. It is desirable to be processed comprehensively.

  Next, the audio signal processing method centering on the sound quality improvement processing of the compressed audio file F in the audio signal processing device 12 of the audio reproduction device 20 shown in FIG. 1 is a flow from step S1 to step S6 as follows. Done in. In particular, the processes characteristic of the present invention are steps S3 and S4.

  Step S1: In FIG. 1, an HD in which a compressed audio file F in which a digital audio signal is compressed with MP3 or WMA is recorded is driven by the HDD 1, and the compressed audio file F is read out and input to the system LSI 10 The MP3 / WMA decoder 11 performs a decoding process.

  Step S2: A process of extracting the bit rate of the compressed audio file F is performed. This can be understood by data analysis in the MP3 / WMA decoder 11.

  Step S3: Harmonic component addition processing for adding a harmonic component lost due to information compression based on the parameter values of the harmonic component generation bandwidth parameter P1 and the harmonic component addition level parameter P2, or by information compression The peak component addition processing for correcting the peak value based on the parameter value of the peak component correction level parameter P3, or the attack component lost by the information compression, the attack component generation bandwidth parameter P4 and the attack component addition level parameter P5 For the sound quality enhancement process that is at least one of the three addition processes of the attack component addition process to be added based on the value, the parameter values of the parameters P1 to P5 are determined in a plurality of stages (large , Medium, Small, None) of the compressed audio file F Rate (low, medium, high) performs processing for automatically setting in accordance with the. This is because the combinations described in [Table 1] to [Table 4] are recorded in advance in a memory (not shown) in the system LSI 10 or the flash ROM 2 and the bits of the compressed audio file F obtained in step S2. An algorithm is determined by the microcomputer program so that the values of the parameters P1 to P5 are selected in consideration of the rate.

  Step S4: The sound signal processing device 12 performs each sound quality improvement process based on the parameter values of the selected and set parameters P1 to P5 of the sound quality improvement process. As a result, as shown in [Table 1], appropriate sound quality enhancement processing corresponding to the bit rate of the compressed audio file F is performed.

  In the flowchart of FIG. 4, in addition to the sound quality enhancement processing method according to the present invention up to step S4, the following known audio signal processing steps S5 and S6 are executed.

  Step S5: D / A-convert the digital audio signal of the compressed audio file F ′ that has been subjected to high sound quality processing, which is the same basic analog conversion processing as before. This includes the digital amplification processing by the preceding stage digital amplifier (class D amplifier).

  Step S6: The DA-converted analog audio signal out is output. This includes output to a stereo speaker output terminal or output to an earphone jack.

  As described above, the audio signal processing apparatus 12 in FIG. 1 of the present invention and the audio signal processing method shown in the flowchart of FIG. 4 are compressed audio files such as MP3 and WMA (the compression method is not limited to these). Depending on various bit rates of F (for example, three levels of low, medium, and high as in the above-described embodiment), the degree of the effect of the high-quality sound processing is several stages (◎ = “more realistic”, ○ = “standard”, △ = “more natural” etc.), and the sound quality enhancement processing is executed. In other words, the sound signal processing form to be called “active sound quality improvement processing method” Particularly, as a portable audio player, it is very easy to handle and has improved stable sound quality.

It is a block diagram centering on the audio | voice signal processing apparatus which concerns on this invention in an audio reproduction apparatus. It is a block diagram of the sound quality improvement process of the audio | voice signal processing apparatus which concerns on this invention in an audio reproduction apparatus. It is a wave form diagram before and behind the sound quality improvement process for demonstrating the outline | summary of the three sound quality improvement processes which concern on this invention. It is a flowchart of the audio | voice signal processing method which concerns on this invention.

Explanation of symbols

1 Hard disk driver
2 Flash ROM
3 SDRAM
4 Liquid crystal display (LCD)
5 Key input device
6 DA converter
7 Output terminal jack
8 USB connector
10 System LSI
11 MP3 / WMA decoder
12 Audio signal processor
13 Sound effect processing device
20 Audio playback device
21 Harmonic component addition processing section
22 Peak component addition processing section
23 Attack component addition processing section
25 High-quality sound processing controller
26 Memory device
F Compressed audio file
F 'High quality audio compressed audio file
P1 Harmonic component generation bandwidth parameter
P2 Harmonic component additional level parameter
P3 Peak component correction level parameter
P4 Attack component generation bandwidth parameter
P5 Attack component additional level parameter

Claims (3)

For audio signals that have been decoded from compressed audio files that are compressed from digital audio signals,
Harmonic component addition to add newly generated harmonic components based on each parameter value of harmonic component generation bandwidth parameter and harmonic component addition level parameter to adjust harmonic components lost by information compression Peak component addition processing that adds a newly generated peak component based on the parameter value of the peak component correction level parameter to adjust the peak value that has slowed down due to processing or information compression, or the attack that was lost due to information compression At least one of the three addition processes of the attack component addition process for adding a newly generated attack component based on the parameter values of the attack component generation bandwidth parameter and the attack component addition level parameter to adjust the component High quality sound processing that is one additional processing,
High quality sound processing means for selecting and setting a parameter value of each parameter from a plurality of predetermined values according to a bit rate of the compressed audio file detected from the compressed audio file. An audio signal processing device. For audio signals that have been decoded from compressed audio files that are compressed from digital audio signals,
Harmonic component addition to add newly generated harmonic components based on each parameter value of harmonic component generation bandwidth parameter and harmonic component addition level parameter to adjust harmonic components lost by information compression Loss due to processing unit or peak component addition processing unit that adds a peak component newly generated based on the parameter value of the peak component correction level parameter to adjust the peak value blunted by information compression or information compression In order to adjust the attack component, an attack component addition processing unit that adds a newly generated attack component based on the parameter values of the attack component generation bandwidth parameter and the attack component addition level parameter, A high quality sound processing unit including at least one additional processing unit, and the compressed audio file Depending on the bit rate of the compressed audio file is detected from,
The parameter value in the parameter of the additional processing unit provided in the high sound quality processing unit is selected and set from a plurality of predetermined values, and the high sound quality processing unit performs the high sound quality processing. An audio signal processing apparatus comprising: a high-quality sound processing control unit for controlling. Inputting and decoding a compressed audio file obtained by compressing information of a digital audio signal;
Extracting the bit rate of the compressed audio file;
Harmonic component addition to add newly generated harmonic components based on each parameter value of harmonic component generation bandwidth parameter and harmonic component addition level parameter to adjust harmonic components lost by information compression Peak component addition processing that adds a newly generated peak component based on the parameter value of the peak component correction level parameter to adjust the peak value that has slowed down due to processing or information compression, or the attack that was lost due to information compression At least one of the three addition processes of the attack component addition process for adding a newly generated attack component based on the parameter values of the attack component generation bandwidth parameter and the attack component addition level parameter to adjust the component About the high quality sound processing that is one additional processing, it is detected from the compressed audio file. A step of said selecting from among a plurality of values are predetermined parameter values of each parameter is set according to the bit rate of the compressed audio file,
Performing the sound quality enhancement process based on the parameter value of each parameter of the set sound quality enhancement process,
An audio signal processing method comprising:

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