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2012-2013年第二学期译,计算机音乐课程期末作业。 Translated in July 2013, for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:白容 Translated in July 2013, drafted by Bai Rong for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2012-2013年第二学期译,计算机音乐课程期末作业。 Translated in June 2013, for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:宋思男 Translated in July 2013, drafted by Song Sinan for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2013年7月译 Translated in July 2013 注意:泛音与分音之间的区别 Note: Please pay attention to difference between Overtone and Partial.
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:王策 Translated in July 2013, drafted by Wang Ce for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:白容 Translated in July 2013, drafted by Bai Rong for the second term of 2012-2013, SYCM, the final assignment of Computer Music course. 注意:此词为双译 Note: "Feedback" has two translations in Chinese
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:周鼎杰 Translated in July 2013, drafted by Zhou Dingjie for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2012-2013年第二学期译,计算机音乐课程期末作业。 初稿:赵其雯 Translated in July 2013, drafted by Zhao Qiwen for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
第一版译于2013年8月,为撰写《恒》作品介绍。 1st Edition in Aug 2013, translated for Heng's introduction
录音:预先录制 Recording: Pre-recorded (Fixed media) 2012-2013年第二学期译,计算机音乐课程期末作业。 Translated in June 2013, for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
乐器数字接口:MIDI MIDI: Musical Instrument Digital Interface 2012-2013年第二学期译,计算机音乐课程期末作业。 Translated in June 2013, for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
Techno 据维基百科:既可译为“高科技舞曲”也可翻译为“铁克诺”。 2012-2013年第二学期译,计算机音乐课程期末作业。 Translated in June 2013, for the second term of 2012-2013, SYCM, the final assignment of Computer Music course.
2002年首版,是世界上第一部用英文撰写的、系统梳理西方音乐理论学术史的鸿篇巨制,该书主编、美国芝加哥大学教授托马斯·克里斯坦森(Thomas Christensen)等32位撰稿人均是世界顶级专家。
这本书是目前西方音乐史书籍中最新,最具时代性的论著之一,正如本书的书名所提示我们的,作者并没有把音乐作为纯粹的专业化的,独立的文化现象加以讨论,而是在广阔的文化背景之下,深入探索音乐和社会文化之间的关系。 节选自译者序
This book is an abridged and slightly altered edition of A History of Music in Western Culture(Prentice Hall 2003). Like the original text, It rests on the premise that the history of music is best conveyed by focusing on a carefully selected repertory of musical works. Once familiar with a representative body of music, students can better grasp the requisite names, dates, and concepts of music history, including an understanding of evolution of musical styles and music's changing uses within the Western tradition. Even more importantly, students will gain a sound basic from which to explore other musical works and repertories.
MySQL (beta) at CHEARSdotinfo.co.uk Additive synthesis is a class of sound synthesis techniques based on the summation of elementary waveforms to create a more complex waveform. Additive synthesis is one of the oldest and most heavily researched synthesis techniques. Additive synthesis has been used since the earliest days of electrical and electronic music (Cahill 1897; Douglas 1968; die Reihe 1955; Stockhausen 1964). The massive Telharmonium synthesizer unveiled in 1906 summed the sound of dozens of electrical tone generators to create additive tone complexes (figure 4.12). Any method that adds several elementary waveforms to create a new one could be classified as a form of additive synthesis. Fixed-waveform Additive Synthesis Some software packages and synthesizers let the musician create waveforms by harmonic addition. In order to make a waveform with a given spectrum the user adjusts the relative strengths of a set of harmonics of a given fundamental. (The term "harmonic" as an integer multiple of a fundamental frequency was first used by Sauveur [1653 1716] in 1701.)
MySQL (beta) at CHEARSdotinfo.co.uk Filter Banks and Equalizers A filter bank is a group of filters that are fed the same signal in parallel (figure 5.30). Each filter is typically a narrow bandpass filter set at a specific frequency. The filtered signals are often combined to form the output sound. When each filter has its own level control the filter bank is called a spectrum shaper. A graphic equalizer has controls that mirror the shape of the filter's frequency response curve (figure 5.31a). Each filter has a fixed center frequency, a fixed bandwidth (typically one-third of an octave), and a fixed Q. (Some units can switch between several Q settings.) The response of each filter can be varied by means of a linear fader to cut or boost specific frequency bands. The potential frequency response of such a filter is shown in figure 5.31b. A parametric equalizer involves a fewer number of filters, but the control of each filter is more flexible. A typical arrangement is to have three or four filters in parallel. Users can adjust independently the center frequency, the Q, and the amount of cut or boost of each filter. A semiparametric equalizer has a fixed Q. A filter that has several regular sharp curves in its frequency response is called a comb filter. The final filter to mention is an allpass filter. For a steady-state (unchanging) sound fed into it, an allpass filter passes all frequencies equally well with unity gainhence its name.All filters introduce some phase shift while attentuating or boosting certain frequencies, but the main effect of an allpass filter is to shift phase. Time-varying Subtractive Synthesis Filters can be fixed or time-variant. In a fixed filter, all the properties of the filter are predefined and do not change over time. This situation is typical of conventional music recording where the sound engineer sets the equalization for each channel at the beginning of the piece. Time-variant filters have many musical applications, particularly in electronic and computer music where the goal is to surpass the limits of traditional instruments. A bandpass filter whose Q, center frequency, and attenuation change over time can impose a enormous variety of sound colorations, particularly if the signal being filtered is also time-varying. A prime example of a system for time-varying subtractive synthesis is the SYTERa digital signal processor developed in the late 1970s at the Groupe de Recherches Musicale (GRM) studio in Paris by Jean-François Analysis/resynthesis systems based on subtractive filters rather than on additive oscillators are capable of approximating any sound. In practice, most of the analysis and data reduction techniques employed in subtractive analysis/resynthesis are geared toward speech synthesis, since this is where most of research has been concentrated (Flanagan et al. 1970; Flanagan 1972). The Vocoder The original subtractive analysis/synthesis system is the vocoder, demonstrated by a talking robot at the 1936 World's Fair in New York City. In musical applications the separation of the driving functions (or resonance) from the excitation function means that rhythm, pitch, and timbre are independently controllable. For example, a composer can change the pitch of a singing voice (by changing the frequency of the excitation function), but retain the original spectral articulation of the voice. By stretching or shrinking the driving functions over time, a piece of spoken text can be slowed down or sped up without shifting the pitch or affecting the formant structure.
MySQL (beta) at CHEARSdotinfo.co.uk Filter Types and Response Curves The specifications of audio equipment usually include a figure for "frequency response." This term is a shorter form of amplitude-versus-frequency response. Each type of filter has its own characteristic frequency response curve. Typical frequency response curves for four basic types of filters are shown in figure 5.23: lowpass, highpass, bandpass and bandreject or notch. Shelving filters, shown in figure 5.24, boost or cut all frequencies above or below a given threshold.Their names can be confusing,because a high shelving filter acts like a lowpass filter when it is adjusted to cut high frequencies,and a low shelving filter acts like a highpass filter when it is adjusted to cut low frequencies. An important property of a filter is its cutoff frequency. The steepness of a filter's slope is usually specified in terms of decibels of attenuation or boost per octave, abbreviated "dB/octave." For example, a 6 dB/octave slope on a lowpass filter makes a smooth attenuation (or rolloff), while a 90 dB/octave slope makes a sharp cutoff (figure 5.26). Filter Q and Gain Many bandpass filters have a control knob (either in software or hardware) for Q. An intuitive definition of Q is that it represents the degree of "resonance" within a bandpass filter.When the Q is high, as in the narrowest inner curve,the frequency response is sharply focused around a peak (resonant) frequency. Q = Freq. center/(Freq. highcutoff - Ffreq. lowcutoff) Another property of a bandpass or bandreject filter is its gain. This is the amount of boost or cut of a frequency band.It shows up as the height (or depth) of the band in a response curve (figure 5.28). When passing a signal through a high Q filter,care must be taken to ensure that the gain at the resonant frequency (the height at the peak) does not overload the system, causing distortion. Many systems have gain-compensation circuits in their filters that prevent this kind of overload.
MySQL (beta) at CHEARSdotinfo.co.uk Introduction to Filters A filter can be literally any operation on a signal (Rabiner et al.1972)! But the most common use of the term describes devices that boost or attenuate regions of a sound spectrum, which is the usage we take up here. Such filters work by using one or both of these methods: · Delaying a copy of an input signal slightly (by one or several sample periods) and combining the delayed input signal with the new input signal (figure 5.21a) · Delaying a copy of the output signal and combining it with the input signal (figure 5.21b) Although figure 5.21 shows combination by summation (+), the combination can also be by subtraction (-). In either case, the combination of original and delayed signals creates a new waveform with a different spectrum. By inserting more delays or mixing sums and differences in various combinations, one can construct a wide range of filter types.
I am Frank (Tianyang Yang). I was a student of Prof. Ruibo Zhang (Mungo) at Shenyang Conservatory of Music from 2010 to 2014. In 2012, I am so glad to become one of the members of CHEARS. Now, I am doing MA degree in Composition for Film and TV at University of Bristol, UK. In June, I attended the EMS 2015 Conference in Sheffield and listened to the presentation of CHEARS in person. Although I could not totally understand the database principle of the CHEARS machine translation, I realize it is the effective tool to reduce the workload of human editing. It will become a system; in other words, the completed materials in database could establish a standard for further translation work within CHEARS machine translation. However, as what Mungo said in EMS 2015, it is just an instrument for CHEARS researchers and the final translation has to be revised by us. In this case, the better human resource management and recruitment could be helpful to our project in the future. I hope, due to our endeavor, CHEARS could narrow the gap of Electroacoustic study between China and UK.