CN101179278B - Sound system and method for encoding sound signals thereof - Google Patents
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Abstract
Description
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/
声音讯号输入端 110Audio
输入缓冲单元 120
编码单元 130
辅助运算单元 131Auxiliary
主要运算单元 133
输出缓冲单元 140
声音讯号输出端 150Audio
模拟数字转换器 20Analog to
数字模拟转换器 30Digital to
数字储存媒体 40
播放单元 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
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
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
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
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/
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
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/
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
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
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
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
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
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
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
且and
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.
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 And take complex conjugate functions f1 and f2 to form conjugate functions f1 * and f2 * , as in step S530.
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产生还原转移函数F1*、F2*,如步骤S550。Again take the complex conjugate and rotate Generate reduction transfer functions F1 * , F2 * , as in step S550.
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.
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.
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