[go: up one dir, main page]
More Web Proxy on the site http://driver.im/

CN101179278B - Sound system and method for encoding sound signals thereof - Google Patents

Sound system and method for encoding sound signals thereof Download PDF

Info

Publication number
CN101179278B
CN101179278B CN2006101439454A CN200610143945A CN101179278B CN 101179278 B CN101179278 B CN 101179278B CN 2006101439454 A CN2006101439454 A CN 2006101439454A CN 200610143945 A CN200610143945 A CN 200610143945A CN 101179278 B CN101179278 B CN 101179278B
Authority
CN
China
Prior art keywords
cosine
signal
sine
function
digital
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Fee Related
Application number
CN2006101439454A
Other languages
Chinese (zh)
Other versions
CN101179278A (en
Inventor
萧诗骏
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Ali Corp
Original Assignee
Ali Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Ali Corp filed Critical Ali Corp
Priority to CN2006101439454A priority Critical patent/CN101179278B/en
Publication of CN101179278A publication Critical patent/CN101179278A/en
Application granted granted Critical
Publication of CN101179278B publication Critical patent/CN101179278B/en
Expired - Fee Related legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Images

Landscapes

  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

A sound system and a method for coding sound signals thereof are provided, the sound signal coding system includes a sound signal input terminal, an input buffer unit, a coding unit, an output buffer unit and a sound signal output terminal. The encoding unit further includes an auxiliary operation unit and a main operation unit. The system shares the operation amount of the main operation unit to encode the voice signal with the auxiliary operation unit, and operates the voice signal with the modulation filter bank, the fast Fourier transform, the improved digital cosine transform and the improved digital sine transform, so as to reduce the complexity of the algorithm, improve the efficiency of processing the voice signal, reduce the whole clock pulse rate and save the power consumption.

Description

音响系统及其声音讯号编码的方法 Sound system and method for encoding sound signals thereof

技术领域technical field

本发明涉及一种音响装置,特别是指一种音响系统及其声音讯号编码的方法。The invention relates to an audio device, in particular to an audio system and a method for encoding audio signals thereof.

背景技术Background technique

在传统的音响装置是以磁带作为录音媒体,然而磁带的体积大、不易保存、录音与放音的音质容易失真,且无法随机存取。In traditional audio devices, tapes are used as recording media. However, tapes are bulky and difficult to store. The sound quality of recording and playback is easily distorted, and random access is impossible.

为了解决上述传统的音响装置的缺点,目前市面上已经发展出以数字媒体作为存取声音讯号的存取媒体,例如记忆卡、随身碟、光盘片等。利用数字储存媒体来存取声音讯号,在运算处理声音讯号上及存取声音讯号上,皆采用数字化的方式,对于存取数字音源所提供的声音讯号甚至可以达到不失真(不压缩)的录音及不失真的放音,或是极少失真(需经过压缩)的录音及极少失真的放音。此外,以数字储存媒体来存取声音讯号的音响装置可以拥有较长的录音时间,且在录音速度上,若是采用数字音源作为录音来源,只要数字音源可以高倍速播放,音响装置便可以达到高倍速录音。In order to solve the shortcomings of the above-mentioned traditional audio devices, digital media such as memory cards, flash drives, and optical discs have been developed on the market as access media for accessing audio signals. Use digital storage media to access audio signals. Digital methods are used for computing and accessing audio signals. For accessing audio signals provided by digital audio sources, undistorted (non-compressed) recordings can be achieved. And undistorted playback, or very little distortion (need to be compressed) recording and very little distortion playback. In addition, audio devices that use digital storage media to access audio signals can have a longer recording time, and in terms of recording speed, if a digital audio source is used as a recording source, as long as the digital audio source can be played at a high speed, the audio device can reach high speed. Double speed recording.

然而,发展以数字储存媒体来存取声音讯号的音响装置,其实现声音讯号压缩的方式是一重要关键。一般而言,目前的音响装置皆以单一数字讯号微处理器(digital signal micro-processor)来作声音讯号压缩上的运算处理。当音响装置处理实时(real time)声音讯号压缩时,若是声音讯号输入速率高到超过数字讯号微处理器的处理数据量的极限,容易造成数据吞吐量不足的现象。However, in the development of audio devices that use digital storage media to access audio signals, the way to achieve audio signal compression is an important key. Generally speaking, current audio devices use a single digital signal microprocessor (digital signal micro-processor) to perform arithmetic processing on audio signal compression. When the audio device processes real time audio signal compression, if the input rate of the audio signal is too high to exceed the limit of the amount of data processed by the digital signal microprocessor, it is easy to cause insufficient data throughput.

发明内容Contents of the invention

本发明的目的是提供一种音响系统及其声音讯号编码的方法,该系统以辅助运算单元来分担主要运算单元编码声音讯号的运算量,并且以调变滤波器排、快速傅立叶转换、改良式数字余弦转换及改良式数字正弦转换来运算声音讯号,以进一步作编码的动作,如此一来,可以降低算法的复杂度、提升处理声音讯号的效率、降低整体时钟脉冲速率,以及节省电力的消耗。The object of the present invention is to provide a sound system and a method for encoding sound signals thereof. The system uses an auxiliary computing unit to share the computational load of the main computing unit for encoding sound signals, and uses modulation filter banks, fast Fourier transform, and improved Digital cosine conversion and improved digital sine conversion are used to calculate the sound signal for further encoding. In this way, the complexity of the algorithm can be reduced, the efficiency of processing the sound signal can be improved, the overall clock rate can be reduced, and power consumption can be saved. .

根据本发明所提供的音响系统中,声音讯号编码系统包含声音讯号输入端、输入缓冲单元、编码单元、输出缓冲单元及声音讯号输出端。编码单元更进一步包含辅助运算单元及主要运算单元。According to the audio system provided by the present invention, the audio signal encoding system includes an audio signal input end, an input buffer unit, an encoding unit, an output buffer unit and an audio signal output end. The encoding unit further includes an auxiliary operation unit and a main operation unit.

当声音讯号由声音讯号输入端输入至编码系统,且为数字讯号时,编码系统会先缓冲声音讯号的数据序列,然后再传送至编码单元中。接着,辅助运算单元会接收声音讯号,并利用具有运算规律性的余弦调变滤波器排、快速傅立叶转换、改良式数字余弦转换等,来分摊运算声音讯号,并将运算过程的状态及运算后的声音讯号传送至主要运算单元。主要运算单元则会接着将辅助运算单元输出的声音讯号作进一步的编码动作,并根据辅助运算单元运算声音讯号的状态,来提供控制讯号至辅助运算单元,使辅助运算单元可以再根据此控制讯号来做进一步的运算动作。最后,输出缓冲单元会缓冲编码单元输出的声音讯号,并传送至声音讯号输出端。When the audio signal is input to the coding system from the audio signal input port and is a digital signal, the coding system will first buffer the data sequence of the audio signal, and then send it to the coding unit. Then, the auxiliary calculation unit will receive the sound signal, and use the regular cosine modulation filter bank, fast Fourier transform, improved digital cosine transform, etc. to share and calculate the sound signal, and compare the status of the calculation process and the result of the calculation. The sound signal is sent to the main computing unit. The main computing unit will then further encode the sound signal output by the auxiliary computing unit, and provide a control signal to the auxiliary computing unit according to the state of the sound signal calculated by the auxiliary computing unit, so that the auxiliary computing unit can then follow this control signal for further operations. Finally, the output buffer unit buffers the audio signal output by the encoding unit and sends it to the audio signal output terminal.

为达上述目的,本发明提供一种音响装置,用以将一声音讯号作录制、播放及转档的处理,其包含:一编码/译码单元,用以编码或译码该声音讯号,其中利用一余弦调变滤波器排、一快速傅立叶转换、一改良式数字余弦转换及一改良式数字正弦转换来加速运算处理该声音讯号,其中该快速傅立叶转换用以实现该余弦调变滤波器排;一数字模拟转换器,用以转换经该编码/译码单元译码后的该声音讯号;一数字储存媒体,用以储存该编码/译码单元所提供的经编码后的该声音讯号,或用以提供该声音讯号;以及一播放单元,用以放大并播放经该数字模拟转换器转换的该声音讯号。To achieve the above purpose, the present invention provides an audio device for recording, playing and transcoding an audio signal, which includes: an encoding/decoding unit for encoding or decoding the audio signal, wherein Accelerated computational processing of the sound signal using a cosine modulated filter bank, a fast fourier transform, a modified digital cosine transform and a modified digital sine transform, wherein the fast fourier transform is used to implement the cosine modulated filter row; a digital-to-analog converter for converting the audio signal decoded by the encoding/decoding unit; a digital storage medium for storing the encoded audio signal provided by the encoding/decoding unit , or for providing the audio signal; and a playback unit for amplifying and playing the audio signal converted by the digital-to-analog converter.

为达上述目的,本发明还提供一种声音讯号编码的方法,其包含:输入一声音讯号至一声音讯号编码系统;利用一余弦调变滤波器排、一快速傅立叶转换、一改良式数字余弦转换及一改良式数字正弦转换来将该声音讯号分成多个子频带,其中该快速傅立叶转换用以实现该余弦调变滤波器排;根据将声音讯号分成这些子频带的状态,产生一控制讯号,并根据该控制讯号产生后续子频带;以及编码这些子频带。In order to achieve the above object, the present invention also provides a method for coding a sound signal, which includes: inputting a sound signal to a sound signal coding system; utilizing a cosine modulation filter bank, a fast Fourier transform, and an improved digital Cosine transformation and a modified digital sine transformation to divide the sound signal into a plurality of sub-bands, wherein the fast Fourier transform is used to realize the cosine modulation filter bank; according to the state of dividing the sound signal into the sub-bands, a control signal is generated , and generate subsequent sub-bands according to the control signal; and encode these sub-bands.

通过上述技术特征,本发明所提供的优点在于,利用辅助运算单元来分担声音讯号编码系统中主要运算单元的运算量。Through the above technical features, the present invention provides the advantage of using the auxiliary computing unit to share the computing load of the main computing unit in the audio signal coding system.

本发明所提供的另一优点在于,利用调变滤波器排、快速傅立叶转换、改良式数字余弦转换及改良式数字正弦转换来将声音讯号分成多个子频带,以进一步作编码的动作。Another advantage provided by the present invention is that the audio signal is divided into multiple sub-bands for further encoding by using modulation filter bank, fast Fourier transform, modified digital cosine transform and modified digital sine transform.

本发明所提供的再一优点在于,利用快速傅立叶转换来实现调变滤波器排。Yet another advantage provided by the present invention is that the modulated filter bank is implemented using a fast Fourier transform.

本发明所提供的再一优点在于,利用改良式数字余弦转换及改良式数字正弦转换来辅助快速傅立叶转换来实现调变滤波器排。Another advantage provided by the present invention is that the modified digital cosine transform and the modified digital sine transform are used to assist the fast Fourier transform to realize the modulated filter bank.

本发明所提供的再一优点在于,利用主要运算单元运算量的分担,可以提升处理声音讯号的效率、降低整体时钟脉冲速率,以及节省电力的消耗。Another advantage provided by the present invention is that the efficiency of processing audio signals can be improved, the overall clock pulse rate can be reduced, and power consumption can be saved by utilizing the sharing of the calculation amount of the main calculation unit.

附图说明Description of drawings

图1为本发明内容的音响系统的方块示意图;Fig. 1 is the schematic block diagram of the sound system of content of the present invention;

图2为本发明内容的编码/译码单元的声音讯号编码系统的方块示意图;2 is a schematic block diagram of an audio signal encoding system of an encoding/decoding unit of the present invention;

图3为本发明内容的声音讯号的编码流程图;Fig. 3 is the coding flowchart of the sound signal of content of the present invention;

图4为本发明内容的以余弦调变滤波器排实现声音讯号编码的流程图;Fig. 4 is the flow chart that realizes sound signal coding with cosine modulation filter row of content of the present invention;

图5为本发明内容的以快速傅立叶转换来实现余弦调变滤波器排的流程图。FIG. 5 is a flow chart of implementing a cosine modulated filter bank by fast Fourier transform according to the present invention.

图中符号说明Explanation of symbols in the figure

编码/译码单元   10Encoding/decoding unit 10

声音讯号输入端  110Audio signal input terminal 110

输入缓冲单元    120Input buffer unit 120

编码单元        130Code unit 130

辅助运算单元    131Auxiliary arithmetic unit 131

主要运算单元    133Main computing unit 133

输出缓冲单元    140output buffer unit 140

声音讯号输出端  150Audio signal output port 150

模拟数字转换器  20Analog to Digital Converter 20

数字模拟转换器  30Digital to Analog Converter 30

数字储存媒体    40digital storage media 40

播放单元        50Playback unit 50

具体实施方式Detailed ways

请参考图1所示,用以说明本发明所提供的音响装置的系统架构,其中图1为本发明内容的音响装置的方块示意图。本发明的音响装置1包含一编码/译码单元(encoding/decoding unit)10、一模拟数字转换器(analogy-to-digital converter,ADC)20、一数字模拟转换器30(digital-to-analogy converter,DAC)、一数字储存媒体40及一播放单元50。Please refer to FIG. 1 for illustrating the system architecture of the audio device provided by the present invention, wherein FIG. 1 is a schematic block diagram of the audio device of the present invention. The audio device 1 of the present invention comprises an encoding/decoding unit (encoding/decoding unit) 10, an analog-to-digital converter (analogy-to-digital converter, ADC) 20, a digital-to-analog converter 30 (digital-to-analogy converter, DAC), a digital storage medium 40 and a playback unit 50.

假设输入至音响装置1的声音讯号X为数字讯号,音响装置1会利用编码/译码单元10直接将声音讯号X编码,最后储存至数字储存媒体40,或再经由数字模拟转换器30转换成模拟讯号后,由播放单元50放大并输出。假设输入至音响装置1的声音讯号X为模拟讯号,则音响装置1会先利用模拟数字转换器20将声音讯号X转换成数字讯号,再传送至编码/译码单元10作编码动作,最后,将编码后的声音讯号X储存至数字储存媒体40,或再经由数字模拟转换器30转换成模拟讯号后,由播放单元50放大并输出。Assuming that the audio signal X input to the audio device 1 is a digital signal, the audio device 1 will use the encoding/decoding unit 10 to directly encode the audio signal X, and finally store it in the digital storage medium 40, or convert it into After the analog signal is amplified and output by the playback unit 50 . Assuming that the audio signal X input to the audio device 1 is an analog signal, the audio device 1 will first use the analog-to-digital converter 20 to convert the audio signal X into a digital signal, and then send it to the encoding/decoding unit 10 for encoding. Finally, The encoded audio signal X is stored in the digital storage medium 40 , or converted into an analog signal by the digital-to-analog converter 30 , amplified and output by the playback unit 50 .

当欲播放储存在数字储存媒体40中的声音讯号X时,音响装置1会直接将声音讯号X由数字储存媒体40中撷取出来,经由编码/译码单元10译码后,再经由数字模拟转换器30将数字形式的声音讯号X转换成模拟讯号,最后再由播放单元50将声音讯号X放大并播放出来。When it is desired to play the audio signal X stored in the digital storage medium 40, the audio device 1 will directly extract the audio signal X from the digital storage medium 40, decode it through the encoding/decoding unit 10, and then pass it through digital simulation. The converter 30 converts the digital audio signal X into an analog signal, and finally the audio signal X is amplified and played by the playback unit 50 .

因此,由上述可知,此数字储存媒体40可以是光盘片等,但本发明并不受限于此,凡可以达到将声音讯号X以数字格式储存的储存媒体,皆为本发明的范围。Therefore, it can be known from the above that the digital storage medium 40 can be an optical disc, etc., but the present invention is not limited thereto, and any storage medium capable of storing the audio signal X in a digital format falls within the scope of the present invention.

本发明进一步提出上述编码/译码单元10内之一编码系统来说明,以辅助运算单元辅助数字讯号处理的编码架构,如图2所示,为本发明内容的编码/译码单元的声音讯号编码系统的方块示意图。声音讯号编码系统包含一声音讯号输入端110、一输入缓冲单元(input buffer)120、一编码单元130、一输出缓冲单元(output buffer)140及一声音讯号输出端150。编码单元130更进一步包含一辅助运算单元(auxiliaryoperating unit)131及一主要运算单元(major operating unit)133。The present invention further proposes an encoding system in the above-mentioned encoding/decoding unit 10 for illustration, and uses an auxiliary computing unit to assist the encoding structure of digital signal processing, as shown in FIG. 2 , which is the audio signal of the encoding/decoding unit of the present invention. Block diagram of the coding system. The audio signal encoding system includes an audio signal input end 110 , an input buffer unit (input buffer) 120 , an encoding unit 130 , an output buffer unit (output buffer) 140 and an audio signal output end 150 . The encoding unit 130 further includes an auxiliary operating unit (auxiliary operating unit) 131 and a main operating unit (major operating unit) 133 .

声音讯号输入端110通过输入缓冲单元120,连接于编码单元130中的辅助运算单元131。编码单元130中的主要运算单元133则通过输出缓冲单元140连接于声音讯号输出端150。辅助运算单元131主要用以辅助主要运算单元133作运算,分担主要运算单元133运算处理声音讯号X的运算量(computon)。而主要运算单元133主要用来编码声音讯号X,并根据辅助运算单元131运算声音讯号X的状态来提供控制讯号给辅助运算单元131作进一步的运算。The audio signal input terminal 110 is connected to the auxiliary operation unit 131 in the encoding unit 130 through the input buffer unit 120 . The main operation unit 133 in the encoding unit 130 is connected to the audio signal output terminal 150 through the output buffer unit 140 . The auxiliary calculation unit 131 is mainly used to assist the main calculation unit 133 in calculation, and share the calculation amount (computon) of the main calculation unit 133 in calculating and processing the sound signal X. The main computing unit 133 is mainly used for encoding the audio signal X, and provides a control signal to the auxiliary computing unit 131 for further computing according to the state of the audio signal X computed by the auxiliary computing unit 131 .

请参考图3所示,为本发明内容的声音讯号的编码流程图。当声音讯号X由声音讯号输入端110输入至编码/译码单元10中的编码系统,且为数字讯号时,如步骤S310,编码系统会先缓冲声音讯号X的数据序列,如步骤S320,然后再传送至编码单元130中。Please refer to FIG. 3 , which is a flow chart of the audio signal encoding in the present invention. When the audio signal X is input to the coding system in the encoding/decoding unit 10 from the audio signal input terminal 110, and is a digital signal, as in step S310, the encoding system will first buffer the data sequence of the audio signal X, as in step S320, and then and then sent to the encoding unit 130.

编码单元130中的辅助运算单元131会接收声音讯号X,并利用具有运算规律性的余弦调变滤波器排(cosine modulated filter bank)、快速傅立叶转换(fast fourier transform,FFT)、改良式数字余弦转换(modified digital cosine transform,MDCT)等,来分摊运算声音讯号X,并将运算过程的状态及运算后的声音讯号X传送至主要运算单元133,如步骤S330。The auxiliary operation unit 131 in the encoding unit 130 receives the sound signal X, and uses a cosine modulated filter bank (cosine modulated filter bank), fast Fourier transform (FFT), and improved digital cosine Transform (modified digital cosine transform, MDCT) etc. to apportion the calculation of the sound signal X, and send the state of the calculation process and the calculated sound signal X to the main calculation unit 133, as in step S330.

主要运算单元133则会接着将辅助运算单元131输出的声音讯号X作进一步的编码动作,并根据辅助运算单元131运算声音讯号X的状态,来提供控制讯号至辅助运算单元131,使辅助运算单元131可以再根据此控制讯号来作进一步的运算动作,如步骤S340。最后,输出缓冲单元140会缓冲编码单元130输出的声音讯号X,如步骤S350,并传送至声音讯号输出端150来输出,如步骤S360。The main calculation unit 133 will then further encode the sound signal X output by the auxiliary calculation unit 131, and provide a control signal to the auxiliary calculation unit 131 according to the state of the sound signal X calculated by the auxiliary calculation unit 131, so that the auxiliary calculation unit The 131 can perform further operations according to the control signal, such as step S340. Finally, the output buffer unit 140 buffers the audio signal X output by the encoding unit 130, as in step S350, and sends it to the audio signal output terminal 150 for output, as in step S360.

因此,由上述可知,此主要运算单元133可以为数字讯号处理器(digital signal processing,DSP)等,而辅助运算单元131可以利用程序设计而成的虚拟装置,但本发明并不受限于此,凡可以达到处理运算声音讯号X的机制,皆为本发明的范围。Therefore, it can be seen from the above that the main computing unit 133 can be a digital signal processor (digital signal processing, DSP), etc., and the auxiliary computing unit 131 can be a virtual device designed by a program, but the present invention is not limited thereto , any mechanism that can achieve processing and computing the sound signal X is within the scope of the present invention.

为了更进一步阐述本发明的编码方法,本发明提出一种利用余弦调变滤波器排来实现声音讯号X编码中声音讯号X压缩的动作,如图4所示,为本发明内容的以余弦调变滤波器排实现声音讯号编码的流程图。In order to further illustrate the encoding method of the present invention, the present invention proposes an action of utilizing a cosine modulation filter bank to realize the compression of the sound signal X in the encoding of the sound signal X, as shown in FIG. The flow chart of changing the filter bank to realize the coding of the sound signal.

当声音讯号X输入至图3中的编码单元130时,由于声音讯号X具有多个输出点,因此首先将声音讯号X的输出点分成多个数据向量,分别为X[1]至X[N],如步骤S410。When the audio signal X is input to the encoding unit 130 in FIG. 3, since the audio signal X has multiple output points, the output points of the audio signal X are first divided into multiple data vectors, respectively X[1] to X[N ], as in step S410.

接着,将这些声音讯号X的数据向量X[1]至X[N]分别利用改良式余弦转换(MDCT)及改良式正弦转换(modified digital sine transform,MDST)来分析,以产生余弦转换函数Y1及正弦转换函数Y2,如步骤S420。Then, the data vectors X[1] to X[N] of these sound signals X are respectively analyzed by modified cosine transform (MDCT) and modified digital sine transform (MDST) to generate cosine transform function Y1 and the sine transfer function Y2, as in step S420.

当编码单元130完成将这些声音讯号X的数据向量X[1]至X[N]转换成余弦转换函数Y1及正弦转换函数Y2后,再一次利用第一对照表使余弦转换函数Y1产生向量倍增的效果,以产生余弦倍增函数W1,以及利用第二对照表使正弦转换函数Y2产生向量倍增的效果,以产生正弦倍增函数W2,如步骤S430。其中,第一对照表为一正弦函数的对照表,第二对照表为一余弦函数的对照表,而余弦倍增函数W1为余弦转换函数Y1与第一对照表的相乘,可视为一矩阵函数,正弦倍增函数W2则为正弦转换函数Y2与第二对照表的相乘,亦可视为一矩阵函数。After the encoding unit 130 completes converting the data vectors X[1] to X[N] of the sound signal X into the cosine transformation function Y1 and the sine transformation function Y2, the cosine transformation function Y1 is used again to multiply the vectors generated by the cosine transformation function Y1 to generate the cosine multiplication function W1, and use the second look-up table to make the sine conversion function Y2 generate the effect of vector multiplication to generate the sine multiplication function W2, as in step S430. Wherein, the first comparison table is a comparison table of a sine function, the second comparison table is a comparison table of a cosine function, and the cosine multiplication function W1 is the multiplication of the cosine conversion function Y1 and the first comparison table, which can be regarded as a As a matrix function, the sine multiplication function W2 is the multiplication of the sine transfer function Y2 and the second comparison table, which can also be regarded as a matrix function.

最后,再将余弦倍增函数W1与正弦倍增函数W2相加,产生调变函数Z,如步骤S440,以进一步输出至主要运算单元133执行编码动作,如步骤S450。由于余弦倍增函数W1与正弦倍增函数W2皆为矩阵函数,因此调变函数Z亦为矩阵函数,且为余弦倍增函数W1与正弦倍增函数W2的整合,可以分成多个频带向量Z[1]至Z[N]。Finally, add the cosine multiplication function W1 and the sine multiplication function W2 to generate the modulation function Z, as in step S440, and further output to the main computing unit 133 for encoding, as in step S450. Since the cosine multiplication function W1 and the sine multiplication function W2 are both matrix functions, the modulation function Z is also a matrix function, and is an integration of the cosine multiplication function W1 and the sine multiplication function W2, which can be divided into multiple frequency band vectors Z[1] to Z[N].

为了更进一步阐述本发明中,改良式余弦转换及改良式正弦转换将数据讯号X分解成余弦转换函数Y1及正弦转换函数Y2的步骤,本发明提出一种利用快速傅立叶转换来实现余弦调变滤波器排的动作,如图5所示,其为本发明内容的以快速傅立叶转换来实现余弦调变滤波器排的流程图。In order to further illustrate the steps of the improved cosine transform and the improved sine transform to decompose the data signal X into a cosine transform function Y1 and a sine transform function Y2 in the present invention, the present invention proposes a method of using fast Fourier transform to realize cosine modulation filtering The action of the filter bank is shown in FIG. 5 , which is a flow chart of implementing the cosine modulation filter bank by fast Fourier transform in the present invention.

首先,将输入至图2的编码单元130内的数据讯号X的输出点重新排序,亦即将数据讯号X的数据向量X[1]至X[N]中的输出点重新排序,如步骤S510。将数据讯号X的输出点排序如下:First, reorder the output points of the data signal X input into the encoding unit 130 in FIG. 2 , that is, reorder the output points in the data vectors X[1] to X[N] of the data signal X, as in step S510. Sort the output points of the data signal X as follows:

ff (( kk )) ≡≡ -- ff (( kk ++ 33 44 NN )) 00 ≤≤ kk ≤≤ NN 44 -- 11 ff (( kk -- 11 44 NN )) NN 44 ≤≤ kk ≤≤ NN -- 11

and

Ff (( ii )) == ΣΣ kk == 00 NN -- 11 ff ~~ (( kk )) ·· sinsin [[ ππ 22 NN (( 22 ii ++ 11 )) (( 22 kk ++ 11 )) ]] ,, ii == 00 ,, .. .. .. ,, NN 22 -- 11 ,,

其中,k代表数据讯号X的第k个输出点,N代表数据讯号X的N个数据向量,I代表数据讯号X的第i个数据向量,而F(i)为f(k)的傅立叶转换公式。Among them, k represents the kth output point of the data signal X, N represents the N data vectors of the data signal X, I represents the ith data vector of the data signal X, and F(i) is the Fourier transform of f(k) formula.

利用变量变换使F(2i+1)=F(N-2i-2),并且合并数据讯号X的输出点,以产生复数函数f1及f2,如步骤S520。Make F(2i+1)=F(N-2i-2) by variable transformation, and combine the output points of the data signal X to generate complex functions f1 and f2, as in step S520.

ff 11 == (( ff ~~ (( 22 kk )) -- ff ~~ (( NN -- 22 kk -- 11 )) )) ++ jj ·· (( ff ~~ (( NN 22 ++ 22 kk )) -- ff ~~ (( NN 22 -- 22 kk -- 11 )) )) ,, 00 ≤≤ kk ≤≤ NN 44 -- 11 ,,

ff 22 == (( ff ~~ (( 22 kk )) ++ ff ~~ (( NN -- 22 kk -- 11 )) )) ++ jj ·&Center Dot; (( ff ~~ (( NN 22 ++ 22 kk )) ++ ff ~~ (( NN 22 -- 22 kk -- 11 )) )) ,, 00 ≤≤ kk ≤≤ NN 44 -- 11 ,,

其中,f1为偶函数的复数函数,f2为奇函数的复数函数。Among them, f1 is a complex function of an even function, and f2 is a complex function of an odd function.

旋转

Figure GA20183719200610143945401D00088
并取复数函数f1、f2的共轭复数,形成共轭函数f1*及f2*,如步骤S530。to rotate
Figure GA20183719200610143945401D00088
And take complex conjugate functions f1 and f2 to form conjugate functions f1 * and f2 * , as in step S530.

ff 11 ** (( kk )) == ff 11 ·&Center Dot; ee 22 πjπj NN (( kk ++ 11 88 ))

ff 22 ** (( kk )) == ff 22 ·&Center Dot; ee 22 πjπj NN (( kk ++ 11 88 ))

利用快速傅立叶转换来作运算,产生转移函数F1、F2,如步骤S540。The fast Fourier transform is used for calculation to generate transfer functions F1 and F2, as in step S540.

Fi=(FFT(f*))* F i =(FFT(f * )) *

再一次取共轭复数并旋转

Figure GA20183719200610143945401D00091
产生还原转移函数F1*、F2*,如步骤S550。Again take the complex conjugate and rotate
Figure GA20183719200610143945401D00091
Generate reduction transfer functions F1 * , F2 * , as in step S550.

Ff ** (( ii )) == Ff (( ii )) ·&Center Dot; ee 22 πjπj NN (( ii ++ 11 88 ))

排序计算结果,产生余弦转换函数Y1及正弦转换函数Y2,如步骤S560。Sort the calculation results to generate the cosine transfer function Y1 and the sine transfer function Y2, as in step S560.

YY 11 Ff (( 22 ii )) == ReRe (( Ff ~~ (( ii )) )) Ff (( 22 ii ++ 11 )) == ImIm (( Ff ~~ (( NN 44 -- ii -- 11 )) )) ,, mm == 00 ~~ NN 44 -- 11 ,,

YY 22 Ff (( 22 ii )) == ImIm (( Ff ~~ (( ii )) )) Ff (( 22 ii ++ 11 )) == ReRe (( Ff ~~ (( NN 44 -- ii -- 11 )) )) ,, mm == 00 ~~ NN 44 -- 11 ,,

由上述可知,藉由利用快速傅立叶转换来实现余弦调变滤波器排,当声音讯号X的子频带数量增加,而导致运算量大幅增加时,本发明内容的声音讯号编码系统可以降低运算声音讯号X所应用到的算法的复杂度,除此之外,因为运算量的分担,可以提升处理声音讯号的效率、降低整体时钟脉冲速率,以及节省电力的消耗。From the above, it can be seen that by using the fast Fourier transform to realize the cosine modulation filter bank, when the number of sub-bands of the sound signal X increases, resulting in a significant increase in the amount of calculation, the sound signal coding system of the present invention can reduce the sound signal calculation. The complexity of the algorithm applied by X, in addition, because of the sharing of the calculation load, can improve the efficiency of processing audio signals, reduce the overall clock pulse rate, and save power consumption.

所示附图仅提供参考与说明用,并非用来对本发明加以限制。以上所述,仅为本发明的较佳可行实施例,非因此即局限本发明的专利范围,故举凡运用本发明说明书及图标内容所为的等效结构变化,均同理包含于本发明的范围内,合予陈明。The drawings shown are provided for reference and illustration only, and are not intended to limit the present invention. The above is only a preferred feasible embodiment of the present invention, and does not limit the patent scope of the present invention. Therefore, all equivalent structural changes made by using the description of the present invention and the contents of the icons are all included in the scope of the present invention. Within the scope, agree with Chen Ming.

Claims (14)

1.一种音响装置,用以将一声音讯号作录制、播放及转档的处理,其特征在于,包含:1. An audio device for recording, playing and transcoding a sound signal, characterized in that it includes: 一编码/译码单元,用以编码或译码该声音讯号,其中利用一余弦调变滤波器排、一快速傅立叶转换、一改良式数字余弦转换及一改良式数字正弦转换来加速运算处理该声音讯号,其中该快速傅立叶转换用以实现该余弦调变滤波器排;An encoding/decoding unit for encoding or decoding the sound signal, wherein a cosine modulation filter bank, a fast Fourier transform, a modified digital cosine transform and a modified digital sine transform are used to accelerate the operation processing the sound signal, wherein the fast Fourier transform is used to implement the cosine modulated filter bank; 一数字模拟转换器,用以转换经该编码/译码单元译码后的该声音讯号;a digital-to-analog converter for converting the audio signal decoded by the encoding/decoding unit; 一数字储存媒体,用以储存该编码/译码单元所提供的经编码后的该声音讯号,或用以提供该声音讯号;以及a digital storage medium for storing the encoded audio signal provided by the encoding/decoding unit, or for providing the audio signal; and 一播放单元,用以放大并播放经该数字模拟转换器转换的该声音讯号。A playing unit is used to amplify and play the audio signal converted by the digital-to-analog converter. 2.如权利要求1所述的音响装置,其特征在于,该声音讯号为一数字讯号。2. The audio device according to claim 1, wherein the audio signal is a digital signal. 3.如权利要求1所述的音响装置,其特征在于,进一步包含一模拟数字转换器,用以转换一模拟讯号以产生该声音讯号。3. The audio device as claimed in claim 1, further comprising an analog-to-digital converter for converting an analog signal to generate the audio signal. 4.如权利要求1所述的音响装置,其特征在于,该编码/译码单元包含:4. The audio device according to claim 1, wherein the encoding/decoding unit comprises: 一辅助运算单元,用以接收该声音讯号,并将该声音讯号作初步运算,以产生一状态讯号;以及an auxiliary operation unit for receiving the sound signal and performing a preliminary operation on the sound signal to generate a status signal; and 一主要运算单元,用以根据该状态讯号来提供一控制讯号至该辅助运算单元,并且将经初步运算后的该声音讯号作编码动作;a main computing unit, which is used to provide a control signal to the auxiliary computing unit according to the state signal, and encode the sound signal after preliminary computing; 其中,该辅助运算单元在接收到该控制讯号后,便依据该控制讯号将该声音讯号作更进一步的运算动作。Wherein, after receiving the control signal, the auxiliary operation unit performs further operation on the sound signal according to the control signal. 5.如权利要求4所述的音响装置,其特征在于,该编码/译码单元还进一步包含:5. The audio device according to claim 4, wherein the encoding/decoding unit further comprises: 一输入缓冲单元,用以缓冲该声音讯号,来进一步作运算及编码动作;以及an input buffer unit for buffering the audio signal for further operation and encoding; and 一输出缓冲单元,用以缓冲该主要运算单元所提供的该声音讯号,以进一步作输出动作。An output buffer unit is used for buffering the audio signal provided by the main computing unit for further output. 6.如权利要求1所述的音响装置,其特征在于,该数字储存媒体为一光盘片、一内建记忆装置或一外接记忆装置。6. The audio device according to claim 1, wherein the digital storage medium is an optical disc, a built-in memory device or an external memory device. 7.如权利要求4所述的音响装置,其特征在于,该辅助运算单元在接收到该声音讯号后,即利用该余弦调变滤波器排、该快速傅立叶转换、该改良式数字余弦转换及该改良式数字正弦转换,将该声音讯号分成多个子频带,以产生该状态讯号,并根据该控制讯号来产生后续的状态讯号;而该主要运算单元接收从该辅助运算单元传送而来的该状态讯号以及经初步运算后的该声音讯号,并根据该状态讯号,产生该控制讯号回传至该辅助运算单元,且编码这些子频带。7. The audio device according to claim 4, wherein after receiving the sound signal, the auxiliary computing unit uses the cosine modulation filter bank, the fast Fourier transform, the improved digital cosine transform and The improved digital sine conversion divides the sound signal into multiple sub-bands to generate the state signal, and generates subsequent state signals according to the control signal; and the main computing unit receives the sent from the auxiliary computing unit The state signal and the audio signal after preliminary calculation are generated, and the control signal is generated according to the state signal and sent back to the auxiliary computing unit, and the sub-frequency bands are coded. 8.如权利要求7所述的音响装置,其特征在于,该辅助运算单元将该声音讯号分成该些子频带,利用该改良式数字余弦转换及该改良式数字正弦转换,来分析该声音讯号,以产生一余弦转换函数及一正弦转换函数并根据一第一对照表及一第二对照表,分别计算该余弦转换函数及该正弦转换函数,以产生一余弦倍增函数及一正弦倍增函数再整合该余弦倍增函数及该正弦倍增函数,以将该声音讯号分成这些子频带。8. The audio device according to claim 7, wherein the auxiliary computing unit divides the sound signal into the frequency sub-bands, and uses the improved digital cosine transform and the improved digital sine transform to analyze the sound signal , to generate a cosine transfer function and a sine transfer function and calculate the cosine transfer function and the sine transfer function respectively according to a first comparison table and a second comparison table to generate a cosine multiplication function and a sine multiplication The function then integrates the cosine multiplication function and the sine multiplication function to divide the sound signal into these sub-bands. 9.如权利要求8所述的音响装置,其特征在于,该辅助运算单元产生该余弦转换函数及该正弦转换函数,通过重新配置该声音讯号的多个输出点,并利用该改良式数字余弦转换及该改良式数字正弦转换,来合并这些输出点,以产生多个复数函数;然后该辅助运算单元分别于旋转这些复数函数后,取这些复数函数的共轭复数,以形成多个共轭函数,并利用该快速傅立叶转换,来运算这些共轭函数,以产生这些共轭函数所相对应的转移函数;接着该辅助运算单元取这些转移函数的共轭复数后,旋转这些转移函数的共轭复数,以产生多个还原转移函数;该辅助运算单元再排序这些还原转移函数,以产生该余弦转移函数及该正弦转移函数。9. The audio device according to claim 8, wherein the auxiliary computing unit generates the cosine transfer function and the sine transfer function by reconfiguring a plurality of output points of the sound signal and utilizing the improved digital cosine transformation and the improved digital sine transformation to combine these output points to generate a plurality of complex functions; then the auxiliary operation unit takes the conjugate complex numbers of these complex functions after rotating these complex functions respectively to form a plurality of conjugate function, and use the fast Fourier transform to operate these conjugate functions to generate the transfer functions corresponding to these conjugate functions; then the auxiliary operation unit takes the conjugate complex numbers of these transfer functions, and rotates the joint of these transfer functions yoke complex numbers to generate a plurality of restored transfer functions; the auxiliary operation unit reorders the restored transfer functions to generate the cosine transfer function and the sine transfer function. 10.如权利要求8所述的音响装置,其特征在于,该第一对照表为一正弦函数对照表,而该第二对照表为一余弦函数对照表。10. The audio device according to claim 8, wherein the first lookup table is a sine function lookup table, and the second lookup table is a cosine function lookup table. 11.一种声音讯号编码的方法,其特征在于,包含:11. A method for coding an audio signal, comprising: 输入一声音讯号至一声音讯号编码系统;Inputting an audio signal into an audio signal encoding system; 利用一余弦调变滤波器排、一快速傅立叶转换、一改良式数字余弦转换及一改良式数字正弦转换来将该声音讯号分成多个子频带,其中该快速傅立叶转换用以实现该余弦调变滤波器排;splitting the sound signal into frequency subbands using a cosine modulation filter bank, a fast fourier transform, a modified digital cosine transform, and a modified digital sine transform, wherein the fast fourier transform is used to effectuate the cosine modulation filter bank; 根据将声音讯号分成这些子频带的状态,产生一控制讯号,并根据该控制讯号产生后续子频带;以及generating a control signal based on the division of the sound signal into the sub-bands, and generating subsequent sub-bands based on the control signal; and 编码这些子频带。These subbands are encoded. 12.如权利要求11所述的声音讯号编码的方法,其特征在于,将该声音讯号分成这些子频带的步骤,包含:12. The method for coding an audio signal as claimed in claim 11, wherein the step of dividing the audio signal into these sub-frequency bands comprises: 利用该改良式数字余弦转换及该改良式数字正弦转换,来分析该声音讯号,以产生一余弦转换函数及一正弦转换函数;analyzing the sound signal using the improved digital cosine transform and the improved digital sine transform to generate a cosine transform function and a sine transform function; 根据一第一对照表及一第二对照表,分别计算该余弦转换函数及该正弦转换函数,以产生一余弦倍增函数及一正弦倍增函数;calculating the cosine transfer function and the sine transfer function respectively according to a first comparison table and a second comparison table to generate a cosine multiplication function and a sine multiplication function; 整合该余弦倍增函数及该正弦倍增函数,以产生这些子频带。The cosine multiplication function and the sine multiplication function are integrated to generate the frequency subbands. 13.如权利要求12所述的声音讯号编码的方法,其特征在于,产生该余弦转换函数及该正弦转换函数的步骤,包含:13. The method for coding an audio signal according to claim 12, wherein the step of generating the cosine transfer function and the sine transfer function comprises: 重新配置该声音讯号的多个输出点;Reconfigure multiple output points of the audio signal; 利用该改良式数字余弦转换及该改良式数字正弦转换,来合并这些输出点,以产生多个复数函数;combining the output points using the modified digital cosine transform and the modified digital sine transform to generate a plurality of complex functions; 分别于旋转这些复数函数后,取共轭复数,以形成多个共轭函数;After rotating these complex functions respectively, take conjugate complex numbers to form multiple conjugate functions; 利用该快速傅立叶转换,来运算这些共轭函数,以产生这些共轭函数所相对应的转移函数;Using the fast Fourier transform to operate these conjugate functions to generate transfer functions corresponding to these conjugate functions; 取这些转移函数的共轭复数后,旋转这些共轭复数,以产生多个还原转移函数;以及After taking the complex conjugates of the transfer functions, rotating the complex conjugates to produce a plurality of reduced transfer functions; and 排序这些还原转移函数,以产生该余弦转移函数及该正弦转移函数。The reduced transfer functions are sorted to generate the cosine transfer function and the sine transfer function. 14.如权利要求12所述的声音讯号编码的方法,其特征在于,该第一对照表为一正弦函数对照表,而该第二对照表为一余弦函数对照表。14. The method for coding an audio signal according to claim 12, wherein the first lookup table is a sine function lookup table, and the second lookup table is a cosine function lookup table.
CN2006101439454A 2006-11-07 2006-11-07 Sound system and method for encoding sound signals thereof Expired - Fee Related CN101179278B (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
CN2006101439454A CN101179278B (en) 2006-11-07 2006-11-07 Sound system and method for encoding sound signals thereof

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
CN2006101439454A CN101179278B (en) 2006-11-07 2006-11-07 Sound system and method for encoding sound signals thereof

Publications (2)

Publication Number Publication Date
CN101179278A CN101179278A (en) 2008-05-14
CN101179278B true CN101179278B (en) 2010-09-08

Family

ID=39405393

Family Applications (1)

Application Number Title Priority Date Filing Date
CN2006101439454A Expired - Fee Related CN101179278B (en) 2006-11-07 2006-11-07 Sound system and method for encoding sound signals thereof

Country Status (1)

Country Link
CN (1) CN101179278B (en)

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN107481727B (en) * 2017-06-23 2020-05-29 罗时志 Audio signal processing method and system based on electric tone pitch control
TWI690221B (en) * 2017-10-18 2020-04-01 宏達國際電子股份有限公司 Sound reproducing method, apparatus and non-transitory computer readable storage medium thereof

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1131507C (en) * 1998-03-11 2003-12-17 松下电器产业株式会社 Audio signal encoding device, decoding device and audio signal encoding-decoding device
US20040088160A1 (en) * 2002-10-30 2004-05-06 Samsung Electronics Co., Ltd. Method for encoding digital audio using advanced psychoacoustic model and apparatus thereof
CN1585469A (en) * 2003-08-19 2005-02-23 仁宝电脑工业股份有限公司 Method for processing video and audio signals

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1131507C (en) * 1998-03-11 2003-12-17 松下电器产业株式会社 Audio signal encoding device, decoding device and audio signal encoding-decoding device
US20040088160A1 (en) * 2002-10-30 2004-05-06 Samsung Electronics Co., Ltd. Method for encoding digital audio using advanced psychoacoustic model and apparatus thereof
CN1585469A (en) * 2003-08-19 2005-02-23 仁宝电脑工业股份有限公司 Method for processing video and audio signals

Also Published As

Publication number Publication date
CN101179278A (en) 2008-05-14

Similar Documents

Publication Publication Date Title
Gersho et al. Vector quantization and signal compression
CN101944362B (en) Integer wavelet transform-based audio lossless compression encoding and decoding method
US8930182B2 (en) Voice transformation with encoded information
US20100257174A1 (en) Method for data compression utilizing pattern-analysis and matching means such as neural networks
JP2010537245A (en) Digital content encoding and / or decoding
TW200529548A (en) Adaptive hybrid transform for signal analysis and synthesis
CN102842337A (en) High-fidelity audio transmission method based on WIFI (Wireless Fidelity)
CN102132494A (en) Method and apparatus of communication
JPH1084284A (en) Signal reproducing method and device
CN1866393A (en) Digital audio player and playing method thereof
JP2019529979A (en) Quantizer with index coding and bit scheduling
CN101010729A (en) Method and device for transcoding
CN101179278B (en) Sound system and method for encoding sound signals thereof
US20220006469A1 (en) Encoding apparatus, decoding apparatus, data structure of code sequence, encoding method, decoding method, encoding program, and decoding program
CN102568484B (en) Warped spectral and fine estimate audio encoding
US20060253288A1 (en) Audio coding and decoding apparatus, computer device incorporating the same, and method thereof
US7020603B2 (en) Audio coding and transcoding using perceptual distortion templates
US9165563B2 (en) Coding device, coding method, decoding device, decoding method, and storage medium
CN118136030A (en) Audio processing method, device, storage medium and electronic device
JP2006003580A (en) Device and method for coding audio signal
Knapen et al. Lossless compression of 1-bit audio
CN100538820C (en) Method and device for processing audio data
JP2014021162A (en) Decoding device, decoding method, and program
TW200912892A (en) Method and apparatus of low-complexity psychoacoustic model applicable for advanced audio coding encoders
KR20140079400A (en) Digital acoustic system

Legal Events

Date Code Title Description
C06 Publication
PB01 Publication
C10 Entry into substantive examination
SE01 Entry into force of request for substantive examination
C14 Grant of patent or utility model
GR01 Patent grant
CF01 Termination of patent right due to non-payment of annual fee

Granted publication date: 20100908

Termination date: 20151107

EXPY Termination of patent right or utility model