JP2005003912A - Audio signal encoding system, audio signal encoding method, and program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an audio encoding technique capable of forming a bit stream of good sound quality by suitably suppressing preechoes while maintaining encoding efficiency. <P>SOLUTION: The audio signal encoding system has a frame dividing section (1) which divides an audio input signal to frames of processing units, an auditory psychology arithmetic section (3) which outputs an auditory entropy value by analyzing a frame divided audio input signal, a block length decision section (4) which determines the conversion block length of the frame based on the auditory entropy value and auditory entropy threshold outputted by the auditory psychology arithmetic section, and a filter bank (2) which blocks the frame according to the block length determined by the block length decision section and converts the blocks to frequency spectra. The block length decision section determines the auditory entropy threshold according to the genre of the audio input signal. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to an audio signal encoding technique, and more particularly, to an audio signal encoding technique using a transform coding technique capable of changing the transform block length.
[0002]
[Prior art]
In recent years, high-quality and high-efficiency audio signal encoding technology has been developed for DVD-Video audio tracks, portable audio players using semiconductor memory, HDD, etc., music distribution via the Internet, home servers in home LAN, etc. It is widely used and widely spread, and its importance is increasing.
[0003]
Many of such audio signal encoding techniques perform time-frequency conversion using a conversion encoding technique. For example, in MPEG2-AAC and Dolby Digital (AC-3), a filter bank is configured by a single orthogonal transform such as MDCT, and in MPEG1 Audio Layer 3 (MP3) and ATRAC (MD), subband division such as QMF is performed. A filter bank is configured by connecting filters and orthogonal transforms in multiple stages.
[0004]
In these high-efficiency audio encoding techniques, the compression efficiency is increased by removing the spectral components determined to be masked by performing masking analysis using human auditory characteristics.
[0005]
The masking analysis used in these high-efficiency audio coding techniques is mainly masking by an audible frequency region in silence and frequency masking by a masker in a critical band.
[0006]
Since the signal determined to be undetectable by humans by the masking analysis is mainly a signal in a high frequency range, in general, the quantization error of the high frequency component can be masked even if it becomes a little larger.
[0007]
However, in the transform coding method, in the case of a so-called transient state in which there is a sudden change in the audio input signal, the quantization error of the high-frequency component in the part where the sudden change occurs is the signal immediately before or immediately after the sudden change. Therefore, ringing noise occurs.
[0008]
As a human auditory characteristic, when a loud sound is generated, it becomes difficult to hear the sound immediately before and immediately after that. This is called a time masking effect. The amount of time that can be heard after a loud sound is relatively long, about 100 msec, although there are individual differences. However, the time of the masking effect that works immediately before is as short as about 5 to 6 msec. Therefore, when ringing noise is generated, noise before a loud sound is easily detected. This is a phenomenon generally called pre-echo.
[0009]
Hereinafter, this phenomenon will be described with reference to the drawings.
[0010]
FIG. 10 is an example of an audio input signal that is changing rapidly. FIG. 11 shows an example of an audio signal obtained by encoding and decoding this signal with 2048 samples, which is a conversion unit in the case of a normal block length of MPEG-2 AAC. As shown in the figure, a quantization error in a high frequency region that occurs in a sudden signal change portion affects the entire block.
[0011]
As described above, immediately before the portion where the amplitude changes abruptly, a human cannot sense noise due to the time masking effect. However, assuming that the input signal uses the same 44.1 KHz sampling frequency as the PCM signal used for the music CD, the conversion unit is converted into time, and the time of 2048 samples is 2048 ÷ 44100 × 1000. = About 46.44 ms, so even if noise occurs in the first half of the time, the pre-masking time will be over and human will perceive the pre-echo.
[0012]
As a method for suppressing this, in various audio encoding methods, by detecting a sudden change in the input signal and shortening the transform block length, the quantization error of the high frequency component due to the sudden change is changed. The occurrence of pre-echo is suppressed by making it not reach the immediately preceding portion.
[0013]
FIG. 12 shows a time domain signal when the audio signal shown in FIG. 10 is encoded and decoded with 256 samples, which is a conversion unit in the case of a short block length in MPEG-2 AAC. In this case, the influence of the quantization error in the high frequency region due to the rapid change of the input signal is confined in the 256 sample block in which the change occurs. As before, when this block length is converted to time at the 44.1 KHz sampling frequency, it becomes approximately 5.80 ms, and thus the human cannot substantially perceive this noise due to the premasking effect, and as a result, the pre-echo disappears.
[0014]
In the actual processing, in order to remove aliasing distortion caused by conversion, the conversion is performed by shifting the input sample by 50% of the conversion length after performing conversion in units of conversion trillions, and superimposing the results. Although this procedure is carried out, this procedure is omitted in the figure for convenience of explanation.
[0015]
However, in general, shortening the block length not only reduces the accuracy of masking analysis due to a decrease in frequency resolution, but also increases the normalization coefficient (hereinafter referred to as scale factor) for each frequency band used for quantization by the number of blocks. Therefore, the amount of information consumed there increases, and bits that should be allocated to spectrum information at the time of quantization are consumed for the scale factor, so that the coding efficiency is lowered. As a result, the quantization error cannot be strictly masked particularly at a low bit rate, and noise may be detected more easily than when the block length is long.
[0016]
Therefore, when determining the actual block length, it is necessary to appropriately determine the balance between suppression of pre-echo and noise generated by a decrease in coding efficiency.
[0017]
As this block length selection method, in MPEG Audio encoding schemes such as MP3 and MPEG-2 AAC, an auditory entropy (hereinafter referred to as PE) is calculated for each block, and if the block length is larger than a predetermined PE threshold, a short block is selected. The length is to be selected.
[0018]
Note that an ID3 tag has been proposed as one of data formats for storing meta information (music title, artist name, album name, genre, creation year, copyright information, etc.) regarding such audio encoded data. Yes. The ID3 tag is generally used particularly in the MP3 file, and is added to the end of the MP3 bit stream.
[0019]
Since the display and editing of the ID3 tag is implemented as a function of a general audio encoding / decoding application, the user can easily edit the contents of the ID3 tag.
[0020]
On the other hand, in recent years, research on techniques for analyzing audio signals themselves and automatically extracting meta information such as genres has been actively conducted. In Patent Document 1 below, the music genre of the music signal is determined by comparing the low-frequency analysis result of the music signal with a plurality of low-frequency component patterns obtained in advance for each music genre. In Patent Document 2 below, the music genre of a music signal is determined using a neural network that has learned patterns such as rhythm, tempo, and adjustment for each genre.
[0021]
Patent Document 3 below discloses a technique for determining block length switching by comparing a threshold value with a difference obtained by subtracting a minimum value from a maximum value of PE.
[0022]
Further, in Patent Document 4 below, when the entropy of auditory entropy is not calculated, the input PCM signal is divided into segments shorter than the block length, a sudden change in amplitude is detected, and a sudden change is detected. A method of estimating the quantization noise in the previous segment and the masking value of the previous segment and selecting a short block length when quantization noise that cannot be masked occurs is disclosed.
[0023]
[Patent Document 1]
JP-A-8-37700
[Patent Document 2]
Japanese Patent Laid-Open No. 10-161654
[Patent Document 3]
JP 2000-276198 A
[Patent Document 4]
JP 2001-142493 A
[0024]
[Problems to be solved by the invention]
As described above, for example, in the block length determination method described in the MPEG-2 AAC standard (ISO / IEC 13818-7: 1997), the block length is determined only by comparing the PE for each block with a fixed PE threshold. is doing. However, since this PE threshold actually changes depending on the device implementation and input signal, when a fixed PE threshold is applied in all cases, the block length is often determined incorrectly depending on the music. The sound quality may be deteriorated due to a decrease in efficiency or a pre-echo.
[0025]
For example, if the block length selection of audio signals of all genres is performed using a conventional fixed PE threshold, a short block length is frequently used in music that is drastically changing, such as hard rock. As a result, unmasked noise increases and sound quality deteriorates. On the other hand, in an adult music such as classical music, a pre-echo occurs without an appropriate short block length being selected.
[0026]
The present invention has been devised in view of the above problems, and by appropriately changing the PE threshold depending on the genre of the input audio signal, by enabling appropriate block selection, a block length shorter than necessary can be achieved. An object of the present invention is to provide an audio encoding technique capable of preventing a selection, appropriately suppressing pre-echo while maintaining encoding efficiency, and creating a bit stream with good sound quality.
[0027]
[Means for Solving the Problems]
According to an aspect of the present invention, a frame dividing unit that divides an audio input signal into frames of processing units, an auditory psychological operation unit that analyzes the audio input signal divided into frames and outputs an auditory entropy value, Based on the auditory entropy value and auditory entropy threshold output by the auditory psychological calculation unit, a block length determination unit that determines a conversion block length of the frame, and the frame according to the block length determined by the block length determination unit An audio signal encoding device, wherein the block length determination unit determines the auditory entropy threshold according to a genre of an audio input signal. Provided.
According to another aspect of the present invention, a frame dividing unit that divides an audio input signal into frames of processing units, an auditory psychological operation unit that analyzes the audio input signal divided into frames and outputs an auditory entropy value, Based on the auditory entropy value and the auditory entropy threshold output by the auditory psychological calculation unit, a block length determination unit that determines a conversion block length of the frame, and a block length determined by the block length determination unit, A filter bank that blocks a frame and converts it into a frequency spectrum, and a history storage unit for storing a history of auditory entropy values of each frame output by the auditory psychological operation unit, the block length determination unit The auditory entropy threshold is set according to the history of auditory entropy values stored in the history storage unit. Audio signal encoding apparatus characterized by a constant is provided.
According to still another aspect of the present invention, a frame dividing step of dividing an audio input signal into frames of processing units, an auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value, A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold, and the frame is blocked according to the determined block length, and a frequency spectrum The audio signal encoding method is characterized in that the block length determination step determines the auditory entropy threshold according to the genre of the audio input signal.
According to still another aspect of the present invention, a frame dividing step of dividing an audio input signal into frames of processing units, an auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value, A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold, and the frame is blocked according to the determined block length, and a frequency spectrum And a history storage step of storing a history of auditory entropy values of each frame output by the auditory psychological calculation step in a history storage unit, wherein the block length determination step includes the history storage unit According to the history of auditory entropy values stored in Audio signal encoding method characterized by determining a Toropi threshold is provided.
According to still another aspect of the present invention, a frame dividing step of dividing an audio input signal into frames of processing units, an auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value, A block length determination block that determines a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold, and blocks the frame according to the determined block length, and a frequency spectrum A program for causing a computer to execute a conversion step for converting into a block, wherein the block length determination step determines the auditory entropy threshold according to a genre of an audio input signal. The
According to still another aspect of the present invention, a frame dividing step of dividing an audio input signal into frames of processing units, an auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value, A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold, and the frame is blocked according to the determined block length, and a frequency spectrum A program for causing a computer to execute a conversion step of converting into a history storage step of storing a history of auditory entropy values of each frame output by the auditory psychological calculation step in a history storage unit, the block length The determination step is stored in the history storage unit. Depending on the history of the objective entropy value, the program and determines the aural entropy threshold is provided.
[0028]
According to the present invention, in order to determine the auditory entropy threshold to be referred to when determining the transform block length according to the genre of the audio input signal, it is possible to perform coding processing with good coding efficiency while avoiding the occurrence of pre-echo, Particularly, encoding processing with good sound quality can be realized at a low bit rate.
[0029]
DETAILED DESCRIPTION OF THE INVENTION
(First embodiment)
Hereinafter, the present invention will be described in detail with reference to the drawings.
FIG. 1 is a configuration example of an audio encoding device according to the first embodiment of the present invention.
[0030]
In the illustrated configuration, reference numeral 1 denotes a frame divider that divides an audio input signal into frames as processing units. The divided frames are sent to a filter bank 2 and an auditory psychological calculator 3 described later.
[0031]
Reference numeral 2 denotes a filter bank, which converts an input time signal divided into frames into a frequency spectrum with a block length of a length designated by the block length determiner 4.
[0032]
An auditory psychological calculator 3 analyzes an audio input signal for each frame, calculates an auditory entropy value, and performs masking calculation for each divided frequency band serving as a quantization unit. As a result of this calculation, the auditory entropy (PE) value is output to the block length determiner 4, and the signal-to-mask ratio (Signal Mask Ratio: SMR) for each divided frequency band is output to the bit allocator 5.
[0033]
Reference numeral 4 denotes a block length determiner, which changes and holds the PE threshold according to genre information sent from the outside, and compares the PE sent from the auditory psychological calculator 3 with the PE threshold to determine the converted block length. , Notify the filter bank 2.
[0034]
A bit allocator 5 determines the bit amount to be allocated to each divided frequency band with reference to the SMR value and frequency spectrum for each divided frequency band transmitted from the auditory psychological calculator 3.
[0035]
Reference numeral 6 denotes a quantizer, which calculates a normalization coefficient (scale factor) of the frequency spectrum output from the filter bank 2 for each frequency band and outputs a frequency according to the bit amount for each frequency band output from the bit allocator 5. Quantize the spectrum.
[0036]
A bit shaper 7 forms a bit stream by appropriately shaping the scale factor and quantized spectrum output from the quantizer 6 into a prescribed format and outputs the bit stream.
[0037]
An audio signal processing operation in the audio signal encoding apparatus having the above configuration will be described below.
[0038]
In the present embodiment, MPEG2-AAC is described as an example of an encoding method for convenience of explanation, but other encoding methods such as MP3 that perform block length switching determination by PE are also realized in exactly the same manner. Is possible.
[0039]
First, prior to processing, each unit is initialized. At this time, the block length determiner 4 changes the PE threshold according to the genre information of the audio signal input from the outside. The PE threshold value can be changed by referring to a table storing PE threshold values for each genre and selecting by genre, or referring to a table storing PE correction values for each genre. A correction value for each genre may be added to the reference value. FIG. 3 shows a configuration example of the PE threshold value table, and FIG. 4 shows a mimic diagram of the PE correction value table. The reference value of PE when the PE correction value table of FIG. In the present embodiment, these tables are stored in the block length determiner 4.
[0040]
An input audio signal such as an audio PCM signal is divided into frames by the frame divider 1 and sent to the filter bank 2 and the psychoacoustic calculator 3. In the case of the MPEG2-AAC LC (Low-Complexity) profile, one frame is composed of 1024 sample PCM signals.
[0041]
The input audio signal is subjected to auditory entropy and masking calculation for each frequency band by the auditory psychological calculator 3 for each frame. The calculated auditory entropy value is compared with the PE threshold value previously changed by the genre information by the block length determiner 4. Here, when the PE value of the frame is larger than the PE threshold, it is determined to use a short block length, and otherwise, it is determined to use a long block length. In the filter bank 2, the input signal is converted into a frequency spectrum with a block length according to this determination.
[0042]
In MPEG2-AAC, in order to eliminate aliasing due to orthogonal transformation in the filter bank 2, overlapping transformation by MDCT is performed. For this reason, in the time-frequency conversion, 2048 samples including the processing target frame and the immediately preceding frame are input as one unit, and 1024 frequency spectra are obtained. At this time, when a long block length is used, 2048 samples of the input signal are orthogonally transformed as one block, and 1024 frequency spectra are output. When a short block length is used, the conversion for outputting 128 frequency spectra with 256 samples of the input signal as one block is performed 8 times while shifting the input signal by 128 samples.
[0043]
Based on the frequency spectrum output from the filter bank 2 and the SMR value output from the psychoacoustic operator 3, the bit allocator 5 allocates bits to each frequency band, and the quantizer 6 determines the scale factor for each frequency band. Calculate and quantize the frequency spectrum according to the bits assigned to each frequency band.
[0044]
The scale factor and the quantized spectrum for each frequency band are shaped into a bit stream according to the format determined by the bit shaper 7 and output.
[0045]
(Second Embodiment)
The second embodiment of the present invention can be implemented as a software program that runs on a general-purpose personal computer (PC). Hereinafter, this case will be described with reference to the drawings.
[0046]
FIG. 5 is a configuration example of an audio signal encoding device according to the second embodiment of the present invention.
[0047]
In the configuration shown in the figure, reference numeral 100 denotes a CPU, which performs operations for audio signal encoding processing, logical determination, and the like, and controls each component connected to the bus via the bus 102.
[0048]
A memory 101 stores a basic I / O program in the configuration example of the present embodiment, a program code being executed, data necessary for program processing, and the like.
[0049]
Reference numeral 102 denotes a bus, which transfers an address signal indicating a component to be controlled by the CPU 100, transfers a control signal of each component to be controlled by the CPU 100, and transfers data between the components. Do.
[0050]
Reference numeral 103 denotes a terminal for instructing device activation, setting of various conditions and input signals, and encoding start.
[0051]
Reference numeral 104 denotes an external storage device, which is an external storage area for storing data, programs, and the like. Data, programs, and the like are stored as necessary, and the stored data and programs are called up when necessary.
[0052]
Reference numeral 105 denotes a media drive. Programs, data, digital audio signals, and the like recorded on the recording medium are loaded into the audio signal encoding apparatus by the media drive 105 reading. In addition, various data and execution programs stored in the external storage device 104 can be written to the recording medium.
[0053]
A microphone 106 collects sound and converts it into an audio signal.
[0054]
Reference numeral 107 denotes a speaker, which can output arbitrary audio signal data as an actual sound.
[0055]
Reference numeral 108 denotes a communication network, which includes a LAN, a public line, a wireless line, a broadcast wave, and the like.
[0056]
Reference numeral 109 denotes a communication interface, which is connected to the communication network 108. The audio signal encoding apparatus according to the present embodiment can communicate with an external device via the communication network via this device and transmit / receive data and programs.
[0057]
In the audio signal encoding apparatus according to the present embodiment including such components, the apparatus operates in response to various inputs from the terminal 103. When an input from the terminal 103 is supplied, an interrupt signal is generated. By being sent to the CPU 100, the CPU 100 reads out various control signals stored in the memory 101, and various controls are performed in accordance with those control signals.
[0058]
The apparatus according to the present embodiment operates when the CPU 100 executes the basic I / O program, the OS, and the audio signal encoding processing program. The basic I / O program is written in the memory 101, and the OS is written in the external storage device 104. When the power of the apparatus is turned on, the OS is read from the external storage device 104 into the memory 101 by the IPL (Initial Program Loading) function in the basic I / O program, and the operation of the OS is started.
[0059]
This audio signal encoding processing program is a program code based on the flowchart of the audio signal encoding processing procedure shown in FIG.
[0060]
FIG. 6 is a content configuration diagram when the audio signal encoding processing program and related data are recorded on a recording medium.
[0061]
In the present embodiment, the audio signal encoding processing program and related data are recorded on a recording medium. As shown in the figure, directory information of the recording medium is recorded in the head area of the recording medium, and thereafter, the audio signal encoding processing program which is the content of the recording medium and the audio signal encoding processing related data are stored. It is recorded as a file.
[0062]
FIG. 7 is a schematic diagram for introducing the audio signal encoding processing program into the audio signal encoding apparatus. The audio signal encoding processing program and related data recorded on the recording medium can be loaded into this apparatus through the media drive 105 as shown in FIG. When the recording medium 110 is set in the media drive 105 of the personal computer, the audio signal encoding processing program and related data are read from the recording medium under the control of the OS and the basic I / O program, and the external storage device 104 is read out. Stored in After that, these information are loaded into the memory 101 at the time of restart and can be operated.
[0063]
FIG. 8 shows a memory map in a state where the audio signal encoding device processing program is loaded into the memory 101 and becomes executable. The memory 101 stores a basic I / O program, an OS, an audio signal encoding processing program, related data, and a work area. At this time, the PE threshold and the PE threshold table shown in FIG. 3 are stored in the work area of the memory 101.
[0064]
Hereinafter, an audio signal encoding process executed by the CPU 100 in the present embodiment will be described according to a flow.
[0065]
FIG. 2 is a flowchart of audio signal encoding processing in the present embodiment.
[0066]
First, step S <b> 1 is a process in which the user designates an input audio signal to be encoded and its genre using the terminal 103. In the present embodiment, the audio signal to be encoded may be an audio PCM file stored in the external storage device 104 or a signal obtained by analog / digital conversion of a real-time audio signal captured by the microphone 106. When the process is finished, step S2 follows.
[0067]
Step S2 is a process of searching the PE threshold value table on the memory 101 and changing the PE threshold value on the memory 101 according to the genre of the audio input signal to be encoded specified in step S1. For example, when the genre input in step S1 is “lock”, 2000 is stored as a PE threshold in a predetermined area on the memory 101 as a result of searching the PE threshold table shown in FIG. When the process is finished, step S3 follows.
[0068]
Step S3 is a process for determining whether or not the input audio signal to be encoded has been completed. If the input signal has ended, the process proceeds to step S12. If not completed, the process proceeds to step S4.
[0069]
Step S4 is a process of dividing the input signal into frames which are processing units for each channel. Similar to the description in the first embodiment, for example, in the case of MPEG2-AAC, each channel is divided into 1024 sample frames. When the process is finished, the process proceeds to step S5.
[0070]
Step S5 is a process of performing auditory psychological calculation of the frame to be encoded. As a result of this calculation, the auditory entropy (PE) of the processing target frame and the SMR value for each divided frequency band that is a quantization unit are calculated. When the process is finished, the process proceeds to step S6.
[0071]
Step S6 is a process of comparing the PE value of the processing target frame calculated in step S5 with the PE threshold stored in step S2 to determine the transform block length. If the PE value of the processing target frame is larger than the PE threshold, the process proceeds to step S7. Otherwise, the process proceeds to step S8.
[0072]
In step S7, based on the determination made in step S6, orthogonal transform using a short block (short block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, eight sets of spectra divided into 128 frequency components are obtained. When the process is finished, the process proceeds to step S9.
[0073]
In step S8, orthogonal transform using a long block (long block length) is performed on the processing target frame based on the determination made in step S6. In the case of MPEG2-AAC, as a result, only one set of spectra divided into 1024 frequency components is obtained. When the process is finished, the process proceeds to step S9.
[0074]
Step S9 is a process of determining the bit amount to be allocated to each divided frequency band from the SMR value for each divided frequency band calculated in step S5, the frequency spectrum obtained in step S7 or step S8, and the encoding bit rate. is there. Such processing is common in the transform coding method as in the present embodiment, and therefore will not be described in detail. When the process is finished, the process proceeds to step S10.
[0075]
Step S10 is a process of calculating a scale factor for each divided frequency band and quantizing the frequency spectrum according to the bit amount assigned in step S9. When the process is finished, the process proceeds to step S11.
[0076]
Step S11 is a process of shaping the scale factor and the quantized spectrum calculated in step S10 according to a format determined by the encoding method and outputting it as a bit stream. In the present embodiment, the bit stream output by this processing may be stored in the external storage device 104, or may be output to an external device connected to the network 108 via the communication interface 109. . When the process is finished, the process proceeds to step S3.
[0077]
Step S12 is a process of shaping the quantized spectrum that has not yet been output due to the delay caused by the psychoacoustic operation or orthogonal transformation, etc., into the bit stream and outputting it. When the process is finished, the audio signal encoding process is finished.
[0078]
As described above, in the audio signal encoding process according to the present embodiment, the PE threshold value to be referred to when determining the block length is appropriately set according to the genre of the audio input signal. A good encoding process is possible, and an encoding process with good sound quality can be realized particularly at a low bit rate. For example, when the genre of the input signal is classic, the PE threshold is set low, so that a short block length is appropriately selected at a portion where a short block length is to be selected, so that the occurrence of pre-echo can be reduced. On the other hand, when the genre is rock, by setting the PE threshold value higher, it is possible to prevent the sound quality from being deteriorated as a result of the excessively short block length being applied and the encoding efficiency being lowered.
[0079]
(Third embodiment)
In the third embodiment of the present invention, the case where the PE threshold is determined by the genre in the ID3 tag input by the user will be described.
[0080]
FIG. 13 is a configuration example of an audio signal encoding device according to the third embodiment of the present invention.
[0081]
As shown in the figure, the audio signal encoding apparatus according to the present embodiment is obtained by adding 8 ID3 tag input devices to the configuration example shown in FIG.
[0082]
Since the constituent elements 1 to 7 are the same as those shown in the first embodiment, the description thereof will be omitted.
[0083]
Reference numeral 8 denotes an ID3 tag input device that accepts input of user ID3 tag information and holds the contents thereof.
[0084]
An audio signal processing operation in the audio signal encoding apparatus having the above configuration will be described below with a focus on differences from the operation of the first embodiment.
[0085]
First, prior to processing, each unit is initialized.
[0086]
Next, an audio input signal to be encoded is designated by the user, and ID3 tag information related to the input signal is input to the ID3 tag input unit 8. At this time, the block length determiner 4 changes the PE threshold according to the input ID3 tag genre. The PE threshold value may be changed by referring to a table storing the PE threshold value for each genre and selecting by the genre number (number) of the ID3 tag, or a table storing the PE correction value for each genre. A search may be made according to the genre number of the ID3 tag, and a correction value for each genre may be added to a value serving as a reference for the PE threshold. FIG. 15 shows a mimic diagram of the PE threshold value table in this case, and FIG. 16 shows a mimic diagram of the PE correction value table. In the present embodiment, the PE reference value when the PE correction value table of FIG. In the present embodiment, these tables are stored in the block length determiner 4.
[0087]
Thereafter, as described above in the first embodiment, the input audio signal is divided by the frame divider 1 and then analyzed by the auditory psychological calculator 3 to calculate the PE value and the SMR. The block length determiner 4 compares the PE value with the PE threshold, and the block length is determined based on the result. The audio input signal that has been divided into frames is converted into a frequency spectrum by the determined block length in the filter bank 2, and the bit allocator 5 applies a bit to each frequency band according to the SMR value output from the psychoacoustic operator 4. The quantizer 6 calculates a normalization coefficient for each frequency band, quantizes the frequency spectrum according to the bits assigned to each frequency band, and converts the frequency spectrum into a bit stream according to the format determined by the bit shaper 7. Formatted and output.
[0088]
When encoding of all input signals is completed, each information of the ID3 tag held in the ID3 tag input unit 8 is added to the end of the bit stream by the bit shaper 7.
[0089]
Hereinafter, a case where similar processing is implemented on a general-purpose PC will be described.
[0090]
The configuration of the audio encoding device shown in FIG. 5 can be implemented in this case as in the second embodiment described above. In the configuration shown in FIG. 5, the PE threshold table shown in FIG. 15 is stored in advance in the work area on the memory 101.
[0091]
Hereinafter, the details of the processing performed in the CPU 100 will be described according to the flow shown in FIG.
[0092]
First, step S <b> 101 is processing in which the user designates an input audio signal to be encoded using the terminal 103. In the present embodiment, the audio signal to be encoded may be an audio PCM file stored in the external storage device 104 or a signal obtained by analog / digital conversion of a real-time audio signal captured by the microphone 106. When the process is finished, step S102 follows.
[0093]
Step S102 is processing in which the user designates the ID3 tag using the terminal 103. As a result of this processing, the ID3 genre No. of the input audio signal is specified as a part of the ID3 tag. When the process is finished, step S 103 follows.
[0094]
Step S103 is a process of searching the PE threshold value table on the memory 101 and changing the PE threshold value on the memory 101 based on the ID3 genre No. of the audio input signal to be encoded specified in step S102. For example, when the ID3 genre No specified in step S102 is “2”, as a result of searching the PE threshold table shown in FIG. 15, 1600 is stored as a PE threshold in a predetermined area on the memory 101. When the process is finished, step S 104 follows.
[0095]
Step S104 is processing for determining whether or not the input audio signal to be encoded has been completed. If the input signal has ended, the process proceeds to step S113. If not completed, the process proceeds to step S105.
[0096]
Step S105 is processing for dividing the input signal into frames, which are processing units, for each channel. Similar to the description in the first embodiment, for example, in the case of MPEG2-AAC, each channel is divided into 1024 sample frames. When the process is finished, the process proceeds to step S106.
[0097]
Step S <b> 106 is processing for performing psychoacoustic calculation of the frame to be encoded. As a result of this calculation, the auditory entropy (PE) of the processing target frame and the SMR value for each divided frequency band that is a quantization unit are calculated. When the process is finished, the process proceeds to step S107.
[0098]
Step S107 is a process of comparing the PE value of the processing target frame calculated in step S106 with the PE threshold stored in step S103 to determine the transform block length. If the PE value of the processing target frame is larger than the PE threshold, the process proceeds to step S108. Otherwise, the process proceeds to step S109.
[0099]
In step S108, based on the determination made in step S107, orthogonal transform using a short block (short block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, eight sets of spectra divided into 128 frequency components are obtained. When the process is finished, the process proceeds to step S110.
[0100]
In step S109, based on the determination made in step S107, orthogonal transform using a long block (long block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, only one set of spectra divided into 1024 frequency components is obtained. When the process is finished, the process proceeds to step S110.
[0101]
Step S110 is a process of determining the bit amount to be allocated to each divided frequency band from the SMR value for each divided frequency band calculated in step S106, the frequency spectrum obtained in step S108 or step S109, and the encoding bit rate. is there. When the process is finished, the process proceeds to step S111.
[0102]
Step S111 is a process of calculating a scale factor for each divided frequency band and quantizing the frequency spectrum according to the bit amount allocated in step S110. When the process is finished, the process proceeds to step S112.
[0103]
Step S112 is processing for shaping the scale factor and the quantized spectrum calculated in step S111 according to a format determined by the encoding method and outputting the result as a bit stream. In this embodiment, the bit stream output by this processing may be stored in an external storage device, or may be output to an external device connected to a line network via a communication interface. When the process is finished, the process proceeds to step S104.
[0104]
Step S113 is a process of shaping the quantized spectrum that has not yet been output due to the delay caused by the psychoacoustic calculation or orthogonal transformation, etc., into the bit stream and outputting it. When the process is finished, the process proceeds to step S114.
[0105]
Step S114 is processing for appropriately encoding and adding the ID3 tag information input in step S102 to the end of the generated encoded bitstream. As a result, when this encoded data is decoded, the ID3 tag information of this data can be accessed, and various information can be used on the decoding processing side. When the process is finished, the audio signal encoding process is finished.
[0106]
As described above, in this embodiment, by inputting an ID3 tag that is standard metadata of audio encoded data, the PE threshold is appropriately changed, so that the user can be changed from a predetermined genre. Since it is sufficient to select the genre, the convenience of the user at the time of encoding is improved, and encoded data with good sound quality can be obtained.
[0107]
(Fourth embodiment)
In the fourth embodiment of the present invention, an example in which the PE threshold is adjusted by the PE history is shown.
[0108]
FIG. 17 is a configuration example of an audio signal encoding device according to the fourth embodiment of the present invention.
[0109]
As shown in the figure, the audio signal encoding apparatus according to this embodiment is obtained by adding nine PE history storage units to the configuration example shown in FIG.
[0110]
Since the constituent elements 1 to 7 are the same as those shown in the first embodiment, the description thereof will be omitted.
[0111]
9 is a PE history temporary storage, and a set of the PE value of each frame output as a result of analyzing the audio input signal by the psychoacoustic operator 3 and the block length determined by the block length determiner 4 as a result, A predetermined number is sequentially stored from a new one. In the present embodiment, the PE history for 100 frames can be stored. A mimetic diagram of the PE history table stored in the PE history temporary storage 9 is shown in FIG.
[0112]
An audio signal processing operation in the audio signal encoding apparatus having the above-described configuration will be described below with a focus on differences from the operation of the first embodiment.
[0113]
In the present embodiment, MPEG2-AAC is described as an example of an encoding method for convenience of explanation, but other encoding methods such as MP3 that perform block length switching determination by PE are also realized in exactly the same manner. Is possible.
[0114]
First, prior to processing, each unit is initialized. At this time, the block length determiner 4 sets an initial PE threshold by referring to the PE threshold table inside the block length determiner 4 according to the genre information of the audio signal input from the outside.
[0115]
An input audio signal such as an audio PCM signal is divided into frames by the frame divider 1 and sent to the filter bank 2 and the psychoacoustic calculator 3.
[0116]
The input audio signal is subjected to auditory entropy (PE) and masking calculation for each frequency band by the psychoacoustic calculator 3 for each frame. The calculated auditory entropy value is sent to the block length determiner 4.
[0117]
The block length determination unit 4 starts correction of the PE threshold value with the input of the PE threshold value as a trigger. The PE threshold correction processing in this embodiment will be described later using the flow of FIG. During this time, the block length determiner 4 can sequentially retrieve the history information necessary for processing by accessing the PE history temporary storage 9.
[0118]
When the correction of the PE threshold is completed, the block length determiner 4 compares the input PE value with the corrected PE threshold. When the PE value of the frame is larger than the PE threshold, it is determined to use a short block length, and when not, it is determined to use a long block length. When the block length is determined, the block length determination unit 4 sends the PE value and the selected block length to the PE history temporary storage unit 9. The PE history temporary storage 9 stores the latest PE and the determined block length in the PE history table. At this time, if the number of history information exceeds a predetermined number, the oldest history is deleted. For example, in the case of the PE history table shown in FIG. 100 rows are deleted from the table, and the PE value and block type as a new history are stored in the top row in the figure. Is shaken again.
[0119]
Thereafter, as described above in the first embodiment, the audio input signal is converted into the frequency spectrum with the determined block length in the filter bank 2, and the bit allocator 5 is output from the psychoacoustic operator 4. A bit is assigned to each frequency band according to the SMR value, and the quantizer 6 calculates a normalization coefficient for each frequency band, quantizes the frequency spectrum according to the bit assigned to each frequency band, and the bit shaper 7 It is formatted into a bitstream according to the specified format and output.
[0120]
Hereinafter, a case where similar processing is implemented on a general-purpose PC will be described.
[0121]
The configuration of the audio encoding device shown in FIG. 5 can be implemented in this case as in the second embodiment described above. In the configuration shown in FIG. 5, the PE threshold value history table shown in FIG. 18 is stored in the work area on the memory 101.
[0122]
Hereinafter, the details of the processing performed in the CPU 100 will be described according to the flow shown in FIG.
[0123]
Step S201 is processing for designating an audio input signal to be encoded by the user and its genre. When the process is finished, step S 202 follows.
[0124]
Step S202 is processing for searching the PE threshold value table and changing the PE threshold value according to the genre specified in step S201. When the process is finished, step S203 follows.
[0125]
Step S203 is a process for determining whether or not the audio input signal is finished. If the input signal has ended, the process proceeds to step S214. If the input signal is not completed, the process proceeds to step S204.
[0126]
Step S <b> 204 is processing for reading an audio input signal for one frame, which is an encoding unit, onto the memory 101. When the process is finished, step S 205 follows.
[0127]
Step S205 is processing for performing auditory psychological calculation of a frame to be encoded. As a result of this calculation, the auditory entropy (PE) of the processing target frame and the SMR value for each divided frequency band that is a quantization unit are calculated. When the process is finished, step S206 follows.
[0128]
Step S206 is processing for comparing the PE value of the processing target frame calculated in step S205 with the PE threshold stored in step S202 to determine the transform block length. If the PE value of the processing target frame is larger than the PE threshold, the process proceeds to step S207. Otherwise, the process proceeds to step S208.
[0129]
In step S207, based on the determination made in step S206, orthogonal transform is performed on the processing target frame using a short block (short block length). In the case of MPEG2-AAC, as a result, eight sets of spectra divided into 128 frequency components are obtained. When the process is finished, the process proceeds to step S209.
[0130]
In step S208, based on the determination made in step S206, orthogonal transform using a long block (long block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, only one set of spectra divided into 1024 frequency components is obtained. When the process is finished, the process proceeds to step S209.
[0131]
Step S209 is a process of determining the bit amount to be allocated to each divided frequency band from the SMR value for each divided frequency band calculated in step S205, the frequency spectrum obtained in step S207 or step S208, and the encoding bit rate. is there. When the process is finished, the process proceeds to step S210.
[0132]
Step S210 is a process of calculating a scale factor for each divided frequency band and quantizing the frequency spectrum according to the bit amount allocated in step S209. When the process is finished, the process proceeds to step S211.
[0133]
Step S211 is processing for shaping the scale factor and the quantized spectrum calculated in step S210 according to a format determined by the encoding method and outputting the result as a bit stream. In the present embodiment, the bit stream output by this processing may be stored in the external storage device 104, or may be output to an external device connected to the network 108 via the communication interface 109. . When the process is finished, the process proceeds to step S212.
[0134]
Step S212 is processing for storing the PE value and selected block of the frame being processed in the PE history table in the memory 101. When the PE history table is full, the oldest history is deleted and the history of this frame is stored. When the process is finished, step S213 follows.
[0135]
Step S213 is processing for correcting the PE threshold based on the history stored in the PE history table. Details of this processing will be described later with reference to FIG. When the process is finished, step S203 follows.
[0136]
Step S214 is a process of shaping the quantized spectrum that has not yet been output due to the delay caused by the psychoacoustic calculation or orthogonal transform, etc., on the memory 101, and outputting it after converting it into a bit stream. When the process is finished, the audio signal encoding process is finished.
[0137]
FIG. 20 is a detailed flow of the PE correction process based on the history in step S213 in the present embodiment.
[0138]
Step S301 is a process of searching for a short block in the PE history table. If there is even one short block in the PE history table, the process proceeds to step S302. Otherwise, the process proceeds to step S311.
[0139]
Step S302 refers to processing for calculating an average PE value (hereinafter, Long_PE) of long blocks remaining in the history with reference to the PE history table. When the process is finished, the process proceeds to step S303.
[0140]
Step S303 is processing for calculating an average PE value of short blocks (hereinafter, Short_PE) with reference to the PE history table. When the process is finished, the process proceeds to step S304.
[0141]
Step S304 is processing to calculate an average PE value, that is, (Long_PE + Short_PE) / 2. In the present embodiment, such calculation is performed as an average PE value because, even in one audio input signal (music), a portion having a high PE value and a portion having a low PE value, that is, a frame to be processed as a short block. This is because the frequency of appearance may be different. In such a case, if the PE average value is the average value of all the frames, the PE average value fluctuates in the middle of the process of encoding the same audio input signal, and the subsequent correction process cannot perform a proper correction. As a result, the sound quality of the created bitstream may not be stable. Therefore, such a calculation method is taken. When the process is finished, the process proceeds to step S305.
[0142]
Step S305 is processing for determining whether or not the average PE value obtained in step S304 is larger than the current PE threshold value. As a result of the determination, if the average PE value is larger, the process proceeds to step S306. Otherwise, the process proceeds to step S307.
[0143]
Step S306 is processing for correcting the PE threshold by adding 10 to the current PE threshold. That is, when the average PE value calculated in step S304 is larger than the current PE threshold value, the PE threshold value is corrected in the plus (+) direction. By doing in this way, PE threshold value can be appropriately corrected with respect to the music in which PE value is changing high. When the process is finished, the process proceeds to step S309.
[0144]
Step S307 is processing for determining whether or not the average PE value obtained in step S304 is smaller than the current PE threshold value. As a result of the determination, if the average PE value is smaller, the process proceeds to step S308. Otherwise, the process proceeds to step S309.
[0145]
Step S308 is processing for correcting the PE threshold by subtracting 10 from the current PE threshold. That is, when the average PE value calculated in step S304 is smaller than the current PE threshold, the PE threshold is corrected in the negative direction. By doing in this way, PE threshold value can be appropriately corrected with respect to music whose PE value transitions low. When the process is finished, the process proceeds to step S309.
[0146]
Step S309 is a process of counting the number of histories that have become short blocks in the PE history table and determining whether or not the number exceeds 20. As a result of the determination, if there are 20 or more short block histories, the process proceeds to step S310. Otherwise, the PE correction process based on the history is terminated and the process returns.
[0147]
Step S310 is processing for correcting the PE threshold by adding 10 to the PE threshold. That is, when there are 20 or more short blocks in the history of 100 frames, it is considered that coding efficiency is lowered due to excessive use of short blocks, and therefore the PE threshold value is corrected in the positive direction. By doing in this way, the control which suppresses the fall of the encoding efficiency by having a low PE threshold value is attained. When the process is finished, the PE correction process based on the history is finished and the process returns.
[0148]
Step S311 is processing for determining whether or not the current PE threshold value has reached the threshold lower limit. As a result of the determination, if the PE threshold value has reached the threshold lower limit, the PE correction process based on the history is terminated and the process returns. Otherwise, the process proceeds to step S312.
[0149]
Step S312 is a process of correcting the PE threshold by subtracting 10 from the PE threshold. In other words, when the history is entirely composed of long blocks, the PE threshold is set higher for the input signal, and there is a possibility that frames that should originally be encoded with short blocks may be overlooked. to correct. However, the correction is performed to such an extent that the preset threshold lower limit is not exceeded. In the case of a quiet song that has almost no change, if the PE threshold value is extremely lowered, the portion that should be encoded with the long block is encoded with the short block, thereby preventing the phenomenon that the encoding efficiency decreases. Because. When the process is finished, the correction process is finished and the process returns.
[0150]
As described above, in the present embodiment, the PE threshold value can be appropriately corrected during processing by using the PE history, so that the PE threshold value corresponding to the input signal can be corrected appropriately. Furthermore, it is possible to perform appropriate block determination, and as a result, it is possible to create a bit stream with better sound quality.
[0151]
(Fifth embodiment)
The fifth embodiment of the present invention shows an example in which an input signal is analyzed to automatically determine a genre.
[0152]
FIG. 21 is a configuration example of an audio signal encoding device according to the fifth embodiment of the present invention.
[0153]
As shown in the figure, the audio signal encoding apparatus according to the present embodiment is obtained by adding ten genre automatic determiners to the configuration example shown in FIG.
[0154]
Since the constituent elements 1 to 7 are the same as those shown in the first embodiment, the description thereof will be omitted.
[0155]
Reference numeral 10 denotes a genre determiner, which analyzes an audio input signal and determines its genre.
[0156]
An audio signal processing operation in the audio signal encoding apparatus having the above configuration will be described below with a focus on differences from the operation of the first embodiment.
[0157]
First, prior to processing, each unit is initialized.
[0158]
Next, an audio input signal to be encoded is designated by the user.
[0159]
Next, the genre determiner 10 analyzes the designated input signal, automatically determines the genre, and sends it to the block length determiner 4. Details of this genre determination method will be described later with reference to FIG. The block length determiner 4 changes the PE threshold according to the received genre. The PE threshold value may be changed by referring to a table storing the PE threshold value for each genre, or a table storing the PE correction value for each genre is searched, and the genre is set as a reference value for the PE threshold value. A correction value for each may be added. FIG. 3 shows a configuration example of the PE threshold value table in this case, and FIG. 4 shows a configuration example of the PE correction value table. In the present embodiment, the PE reference value when the correction value table of FIG. In the present embodiment, these tables are stored in the block length determiner 4.
[0160]
Thereafter, as described above in the first embodiment, the input audio signal is divided by the frame divider 1 and then analyzed by the auditory psychological calculator 3 to calculate the PE value and the SMR. The block length determiner 4 compares the PE value with the PE threshold, and the block length is determined based on the result. The frame-divided audio input signal is converted into a frequency spectrum with the determined block length in the filter bank 2, and the bit allocator 5 is assigned a bit in each frequency band according to the SMR value output from the auditory psychological calculator 3. , The quantizer 6 calculates a normalization coefficient for each frequency band, quantizes the frequency spectrum according to the bits allocated to each frequency band, and converts the frequency spectrum into a bit stream according to the format determined by the bit shaper 7. Formatted and output.
[0161]
Hereinafter, a case where similar processing is implemented on a general-purpose PC will be described.
[0162]
The configuration of the audio encoding device shown in FIG. 5 can be implemented in this case as in the second embodiment described above.
[0163]
Hereinafter, the details of the processing performed in the CPU 100 will be described according to the flow shown in FIG.
[0164]
FIG. 22 is a flow of audio signal encoding processing in the present embodiment.
[0165]
First, step S <b> 401 is processing in which the user designates an input audio signal to be encoded using the terminal 103. In the present embodiment, the audio signal to be encoded may be an audio PCM file stored in the external storage device 104 or a signal obtained by analog / digital conversion of a real-time audio signal captured by the microphone 106. When the process is finished, step S 402 follows.
[0166]
Step S402 is processing for automatically determining the genre of the input audio signal. Details of this processing will be described later with reference to FIG. When the process is finished, step S 403 follows.
[0167]
Step S403 is a process of searching the PE threshold value table on the memory 101 and changing the PE threshold value on the memory 101 according to the genre of the audio input signal to be encoded determined in step S402. When the process is finished, step S 404 follows.
[0168]
Step S404 is processing for determining whether or not the input audio signal to be encoded has been completed. If the input signal has ended, the process proceeds to step S413. If not completed, the process proceeds to step S405.
[0169]
Step S405 is processing for dividing the input signal into frames which are processing units for each channel. Similar to the description in the first embodiment, for example, in the case of MPEG2-AAC, each channel is divided into 1024 sample frames. When the process is finished, the process proceeds to step S406.
[0170]
Step S406 is processing for performing psychoacoustic calculation of the frame to be encoded. As a result of this calculation, the auditory entropy (PE) of the processing target frame and the SMR value for each divided frequency band that is a quantization unit are calculated. When the process is finished, the process advances to step S407.
[0171]
Step S407 is processing for comparing the PE value of the processing target frame calculated in step S406 with the PE threshold stored in step S403 to determine the transform block length. If the PE value of the processing target frame is larger than the PE threshold, the process proceeds to step S408. Otherwise, the process proceeds to step S409.
[0172]
In step S408, based on the determination made in step S107, orthogonal transform using a short block (short block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, eight sets of spectra divided into 128 frequency components are obtained. When the process is finished, the process proceeds to step S410.
[0173]
In step S409, based on the determination made in step S407, orthogonal transform using a long block (long block length) is performed on the processing target frame. In the case of MPEG2-AAC, as a result, only one set of spectra divided into 1024 frequency components is obtained. When the process is finished, the process proceeds to step S410.
[0174]
Step S410 is a process of determining the bit amount to be assigned to each divided frequency band from the SMR value for each divided frequency band calculated in step S406, the frequency spectrum obtained in step S408 or step S409, and the encoding bit rate. is there. Such processing is common in the transform coding method as in the present embodiment, and therefore will not be described in detail. When the process is finished, the process advances to step S411.
[0175]
Step S411 is a process of calculating the scale factor for each divided frequency band and quantizing the frequency spectrum according to the bit amount allocated in step S410. When the process is finished, the process advances to step S412.
[0176]
Step S412 is a process of shaping the scale factor and the quantized spectrum calculated in step S411 according to a format determined by the encoding method and outputting it as a bit stream. In the present embodiment, the bit stream output by this processing may be stored in the external storage device 104, or may be output to an external device connected to the network 108 via the communication interface 109. . When the process is finished, the process proceeds to step S404.
[0177]
Step S413 is a process of shaping the quantized spectrum that has not yet been output due to the delay caused by the psychoacoustic operation or orthogonal transformation, etc., into the bit stream and outputting it. When the process is finished, the audio signal encoding process is finished.
[0178]
FIG. 23 is a flow detailing the genre automatic determination process in step S402 in the present embodiment.
[0179]
Step S501 is processing for extracting the rhythm of the audio input signal by detecting the amplitude of the audio input signal and the repetition pattern. When the process is finished, the process proceeds to step S502.
[0180]
Step S502 is a process of calculating the tempo of the input signal from the rhythm extracted in step S501 and the sampling rate of the input audio signal. When the process is finished, the process advances to step S503.
[0181]
Step S503 is processing for converting an audio input signal into a frequency spectrum using orthogonal transform such as FFT and analyzing a pure tone component. When the process is finished, the process advances to step S504.
[0182]
Step S504 is a process of extracting the melody by analyzing the melody, chord composition, chord progression, etc. of the music based on the frequency analysis result performed in step S503. When the process is finished, the process advances to step S505.
[0183]
Step S505 is a process of comprehensively analyzing the rhythm, tempo, and music tone extracted so far to determine the genre of the input signal. As this determination method, for example, a database storing these features and genres is searched, and a genre is determined by selecting a genre having the most similar pattern, or a feature of music for each genre is learned. There are techniques such as determination using a neural network circuit, but these are all known and will not be described in detail. As a result of this determination, if the genre cannot be determined, the process proceeds to step S506. If the genre can be determined, the process proceeds to step S507.
[0184]
Step S506 is processing for selecting a default genre as genre information. In this embodiment, the default genre is a genre given to an input signal whose genre cannot be determined. When this information is set as a genre, a default value is set as the PE threshold value in step S403 of FIG. The When the process is finished, the process advances to step S507.
[0185]
Step S507 is processing to store the genre information selected in step S505 or step S506 in the work area on the memory 101. When the process is finished, the genre automatic determination process is finished and the process returns.
[0186]
As described above, according to the present embodiment, the user simply specifies the audio input signal, the genre is automatically determined, and the PE threshold corresponding to the genre is appropriately set. It is possible to obtain a bit stream with good sound quality without being applied.
[0187]
(Other embodiments)
In addition, this invention is not limited to embodiment mentioned above.
[0188]
In the second embodiment described above, the PE threshold value is changed according to the genre by searching the PE threshold value table of FIG. 3, but this can also be realized by searching the PE threshold value correction table of FIG. is there. FIG. 9 is a memory map configuration diagram in this case. This memory map is basically the same as the memory map of FIG. 8, but the PE threshold value, the PE reference value, and the PE threshold value correction table are stored in the work area.
[0189]
In the second embodiment described above, the PE threshold value is stored on the memory map, but the same processing can be realized by storing a pointer indicating a value on the PE threshold table instead. .
[0190]
In the above-described embodiment, no particular reference is made to the recording medium, but this is applicable to any recording medium such as FD, HDD, CD, DVD, MO, and semiconductor memory.
[0191]
In the fourth embodiment described above, the PE history table is configured as normal tabular data, but this is applicable even when configured using a ring buffer capable of storing a predetermined number of lists. Is possible.
[0192]
In the fourth embodiment described above, the PE threshold value is corrected every time one frame is processed. However, this correction may be performed at a frequency of several frames or tens of frames.
[0193]
As described above, according to the above-described embodiment, the PE threshold to be referred to when determining the block length is appropriately set according to the genre of the audio input signal. In particular, it is possible to realize encoding processing with good sound quality at a low bit rate. For example, when the genre of the input signal is classic, the PE threshold is set low, so that a short block length is appropriately selected at a portion where a short block length is to be selected, so that the occurrence of pre-echo can be reduced. On the other hand, when the genre is rock, it is possible to prevent a short block length from being applied excessively, resulting in a decrease in encoding efficiency and consequently deterioration in sound quality. In other words, it is possible to select an appropriate block length according to the characteristics of the input signal, and it is possible to create a high-quality bitstream while suppressing the occurrence of pre-echo while preventing a decrease in encoding efficiency.
[0194]
Furthermore, according to the above embodiment, the PE threshold value is appropriately set without making the user aware of it by changing the PE threshold value according to the genre information of the ID3 tag which is metadata generally used in the encoded audio data. Therefore, it is possible to obtain a bit stream with good sound quality without extra user effort. In other words, it is possible to select an appropriate block length according to the genre of the input signal without extra user effort.
[0195]
Furthermore, according to the above-described embodiment, the PE threshold value can be corrected according to the input signal by appropriately correcting the PE threshold value based on the past PE value and the history of block selection information. It is possible to appropriately select a block that could not be adjusted, and a bit stream with better sound quality can be obtained. That is, block length selection can be appropriately controlled according to the input signal, and a higher quality bit stream can be provided.
[0196]
Furthermore, according to the above-described embodiment, by automatically determining the genre of the input signal and thereby changing the PE threshold, it is possible to set the PE threshold according to the genre of the input signal without making the user aware of it. , User convenience is improved.
[0197]
This embodiment can be realized by a computer executing a program. Also, means for supplying a program to a computer, for example, a computer-readable recording medium such as a CD-ROM recording such a program, or a transmission medium such as the Internet for transmitting such a program is also applied as an embodiment of the present invention. Can do. A computer program product such as a computer-readable recording medium in which the above program is recorded can also be applied as an embodiment of the present invention. The above program, recording medium, transmission medium, and computer program product are included in the scope of the present invention. As the recording medium, for example, a flexible disk, a hard disk, an optical disk, a magneto-optical disk, a CD-ROM, a magnetic tape, a nonvolatile memory card, a ROM, or the like can be used.
[0198]
The above-described embodiments are merely examples of implementation in carrying out the present invention, and the technical scope of the present invention should not be construed in a limited manner. That is, the present invention can be implemented in various forms without departing from the technical idea or the main features thereof.
[0199]
【The invention's effect】
As described above, according to the present invention, the auditory entropy threshold value to be referred to when determining the transform block length is determined according to the genre of the audio input signal. Encoding processing is possible, and encoding processing with good sound quality can be realized particularly at low bit rates.
[Brief description of the drawings]
FIG. 1 is a diagram illustrating a configuration example of an audio signal encoding device according to a first embodiment of the present invention.
FIG. 2 is a flowchart of an audio signal encoding process in the second embodiment of the present invention.
FIG. 3 is a schematic diagram of a PE threshold value table according to the first embodiment of the present invention.
FIG. 4 is a schematic diagram of a PE correction value table according to the first embodiment of the present invention.
FIG. 5 is a diagram illustrating a configuration example of an audio signal encoding device according to a second embodiment of the present invention.
FIG. 6 is a diagram showing a content configuration example of a storage medium storing an audio signal encoding processing program according to the second embodiment of the present invention.
FIG. 7 is a schematic diagram for introducing an audio signal encoding process according to the second embodiment of the present invention into a personal computer.
FIG. 8 is a configuration diagram of a memory map in a second embodiment of the present invention.
FIG. 9 is a configuration diagram of a memory map according to another embodiment of the present invention.
FIG. 10 is a mimic diagram of an audio signal.
11 is a mimic diagram of an audio signal when the audio signal shown in FIG. 10 is encoded and decoded in units of 2048 samples. FIG.
12 is a mimic diagram of an audio signal when the audio signal shown in FIG. 10 is encoded and decoded in units of 256 samples.
FIG. 13 is a diagram illustrating a configuration example of an audio signal encoding device according to a third embodiment of the present invention.
FIG. 14 is a flowchart of audio signal encoding processing according to the third embodiment of the present invention.
FIG. 15 is a schematic diagram of a PE threshold value table according to the third embodiment of the present invention.
FIG. 16 is a schematic diagram of a PE correction value table according to the third embodiment of the present invention.
FIG. 17 is a diagram illustrating a configuration example of an audio signal encoding device according to a fourth embodiment of the present invention.
FIG. 18 is a mimic diagram of a PE history table according to the fourth embodiment of the present invention.
FIG. 19 is a flowchart of audio signal encoding processing in the fourth embodiment of the present invention;
FIG. 20 is a flowchart of PE threshold value correction processing according to the fourth embodiment of the present invention.
FIG. 21 is a diagram illustrating a configuration example of an audio signal encoding device according to a fifth embodiment of the present invention.
FIG. 22 is a flowchart of audio signal encoding processing in the fifth embodiment of the present invention;
FIG. 23 is a flowchart of genre automatic determination processing in the fifth embodiment of the present invention;
[Explanation of symbols]
1 Frame divider
2 filter banks
3 auditory psychological calculator
4 Block length detector
5-bit allocator
6 Quantizer
7-bit shaper
8 ID3 tag input device
9 PE history temporary storage
10 Genre automatic judgment device
100 CPU
101 memory
102 bus
103 terminals
104 External storage device
105 media drive
106 microphone
107 Speaker
108 Communication line
109 Communication interface
110 Storage media

Claims (21)

A frame dividing unit that divides the audio input signal into frames of processing units;
An auditory psychological operation unit that analyzes the audio input signal divided into frames and outputs an auditory entropy value;
Based on the auditory entropy value and auditory entropy threshold output by the auditory psychological calculation unit, a block length determination unit that determines a conversion block length of the frame;
According to the block length determined by the block length determination unit, the frame is blocked and converted to a frequency spectrum, and a filter bank,
The audio signal encoding apparatus, wherein the block length determination unit determines the auditory entropy threshold according to a genre of an audio input signal. A bit allocation unit that divides the frequency spectrum converted by the filter bank into a plurality of frequency bands, and determines a bit amount to be allocated to each frequency band according to a signal-to-mask ratio calculated by the auditory psychological calculation unit; ,
A quantization unit that quantizes a frequency spectrum transformed by the filter bank according to the bit allocation determined by the bit allocation unit;
The audio signal encoding apparatus according to claim 1, further comprising: a bit shaping unit that generates a bit stream obtained by shaping the quantized spectrum quantized by the quantization unit according to a prescribed format. The audio signal encoding according to claim 1 or 2, wherein the block length determination unit determines the auditory entropy threshold by adding or subtracting a correction value for each genre to a reference value of the auditory entropy threshold. apparatus. The audio signal encoding apparatus according to claim 1, wherein a genre of the audio input signal is designated by a user. Furthermore, it has an ID3 tag input unit for inputting an ID3 tag,
The audio signal according to any one of claims 1 to 4, wherein the block length determination unit determines the auditory entropy threshold according to a genre of an ID3 tag input by the ID3 tag input unit. Encoding device. Furthermore, a history storage unit for storing a history of auditory entropy values of each frame output by the auditory psychological calculation unit,
The said block length determination part determines the said auditory entropy threshold value according to the log | history of the auditory entropy value preserve | saved at the said log | history preservation | save part, The any one of Claims 1-5 characterized by the above-mentioned. Audio signal encoding device. The history storage unit stores the history of the transform block length together with the history of auditory entropy values,
The audio signal encoding apparatus according to claim 6, wherein the block length determination unit determines the auditory entropy threshold according to the history of the auditory entropy value and the history of the transform block length. 8. The audio signal encoding apparatus according to claim 7, wherein the block length determination unit selects and determines either a long block length or a short block length as the transform block length. The block length determination unit refers to the history stored in the history storage unit, according to the average value of the auditory entropy value when the long block length is selected and the average value of the auditory entropy value when the short block length is selected. 9. The audio signal encoding apparatus according to claim 8, wherein the auditory entropy threshold is determined. The audio signal encoding apparatus according to claim 8 or 9, wherein the block length determination unit determines the auditory entropy threshold according to the number of short block length selections. Furthermore, it has a genre determiner for determining the genre of the audio input signal,
The audio signal encoding apparatus according to claim 1 or 2, wherein the block length determination unit determines the auditory entropy threshold according to the genre determined by the genre determiner. A frame dividing unit that divides the audio input signal into frames of processing units;
An auditory psychological operation unit that analyzes the audio input signal divided into frames and outputs an auditory entropy value;
Based on the auditory entropy value and auditory entropy threshold output by the auditory psychological calculation unit, a block length determination unit that determines a conversion block length of the frame;
A filter bank that blocks the frame according to the block length determined by the block length determination unit and converts it into a frequency spectrum;
A history storage unit for storing a history of auditory entropy values of each frame output by the auditory psychological calculation unit;
The audio signal encoding apparatus, wherein the block length determination unit determines the auditory entropy threshold according to a history of auditory entropy values stored in the history storage unit. A bit allocation unit that divides the frequency spectrum converted by the filter bank into a plurality of frequency bands, and determines a bit amount to be allocated to each frequency band according to a signal-to-mask ratio calculated by the auditory psychological calculation unit; ,
A quantization unit that quantizes a frequency spectrum transformed by the filter bank according to the bit allocation determined by the bit allocation unit;
13. The audio signal encoding apparatus according to claim 12, further comprising: a bit shaping unit that generates a bit stream obtained by shaping the quantized spectrum quantized by the quantization unit according to a prescribed format. The history storage unit stores the history of the transform block length together with the history of auditory entropy values,
The audio signal encoding apparatus according to claim 12 or 13, wherein the block length determination unit determines the auditory entropy threshold according to the history of the auditory entropy value and the history of the transform block length. 15. The audio signal encoding apparatus according to claim 14, wherein the block length determination unit selects and determines either a long block length or a short block length as the transform block length. The block length determination unit refers to the history stored in the history storage unit, according to the average value of the auditory entropy value when the long block length is selected and the average value of the auditory entropy value when the short block length is selected. The audio signal encoding apparatus according to claim 15, wherein the auditory entropy threshold is determined. The audio signal encoding apparatus according to claim 15 or 16, wherein the block length determination unit determines the auditory entropy threshold according to the number of short block length selections. A frame dividing step for dividing the audio input signal into frames of processing units;
An auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value;
A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold;
And converting the frame into a frequency spectrum according to the determined block length,
The audio signal encoding method, wherein the block length determination step determines the auditory entropy threshold according to a genre of an audio input signal. A frame dividing step for dividing the audio input signal into frames of processing units;
An auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value;
A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold;
According to the determined block length, converting the frame into a frequency spectrum;
A history storage step of storing a history of auditory entropy values of each frame output by the auditory psychological calculation step in a history storage unit;
The audio signal encoding method, wherein the block length determining step determines the auditory entropy threshold according to a history of auditory entropy values stored in the history storage unit. A frame dividing step for dividing the audio input signal into frames of processing units;
An auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value;
A block length determination block for determining a transform block length of the frame based on the output auditory entropy value and auditory entropy threshold;
A program for causing a computer to execute a conversion step of blocking the frame according to the determined block length and converting the frame into a frequency spectrum,
The block length determination step determines the auditory entropy threshold according to a genre of an audio input signal. A frame dividing step for dividing the audio input signal into frames of processing units;
An auditory psychological calculation step of analyzing the frame-divided audio input signal and outputting an auditory entropy value;
A block length determination step for determining a transform block length of the frame based on the output auditory entropy value and an auditory entropy threshold;
According to the determined block length, converting the frame into a frequency spectrum;
A program for causing a computer to execute a history storage step of storing a history of auditory entropy values of each frame output by the auditory psychological calculation step in a history storage unit,
The block length determination step determines the auditory entropy threshold according to a history of auditory entropy values stored in the history storage unit.

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