Mason Blake
Computers can generate sound in multiple ways. One of the most common methods is through digital audio conversion (DAC), where computers use digital-analog converters to convert samples of sound into electrical signals that drive speakers. These samples are taken by measuring the height of certain waveforms at specific points in time and are then converted into binary number values. By changing the time between these electrical pulses, higher and lower tones are generated. Computers can also create sound without relying on previous recordings through methods like Frequency Modulation (FM) Synthesis, where multiple sound waves are overlapped, and Wave Table Synthesis, which uses real instrument samples.
| Characteristics | Values |
|---|---|
| Sound Creation | Frequency Modulation (FM) Synthesis, Wave Table Synthesis, Digital Audio Workstations, Oscillator Circuits |
| Sound Recording | Analog to Digital Conversion (ADC), Sampling Rate, Quantization |
| Sound Output | Digital-Analog Converter (DAC), Electric Pulses |
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What You'll Learn
Digital-analog conversion
Computers generate audio by sending electric pulses to a speaker, which then produces sound. The process of digital-analog conversion is integral to this sound creation. This conversion involves taking a continuous-time and continuous-amplitude analog signal and transforming it into a discrete-time and discrete-amplitude digital signal. This process is known as analog-to-digital conversion (ADC).
In the context of sound creation, analog-to-digital conversion involves converting real-world analog audio signals, which vary in amplitude, into digital data that a computer can process. This conversion is achieved through a system called an analog-to-digital converter (ADC). The ADC takes the analog waveform and chops it up into thousands of samples per second, with each sample having a specific value. The more samples taken per second, the more precise the digital representation becomes, capturing the nuances of the original sound. This process is essential for music production, as it allows for the manipulation and sharing of audio in ways not possible with analog signals alone.
Once the analog sound is converted into a digital format, various creative options become available. Digital audio workstations (DAWs) enable editing, layering, and manipulation of sounds. Virtual instruments and effects can be utilized to create complex audio compositions. This digital realm provides a playground for artists and producers to experiment and craft their desired sounds.
However, for the computer to output this digitally processed sound, it needs to be converted back into an analog signal that can be understood by the speakers. This is where digital-to-analog conversion (DAC) comes into play. The digital data is fed into a digital-to-analog converter (DAC), which transforms the digital information into an analog electrical signal. This signal then drives an audio amplifier, which, in turn, powers the speaker, resulting in the production of sound.
DACs are commonly used in music players, televisions, and mobile phones to convert digital data streams into analog audio or video signals. The suitability of a DAC for a specific application depends on factors such as resolution and maximum sampling frequency. It is important to select a DAC with insignificant errors to minimize signal degradation during the conversion process.
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Frequency modulation synthesis
FM synthesis was discovered in 1967 by John Chowning at Stanford University. Chowning initially used complex waveforms to modulate the pitch of simple sine waves. He then began experimenting with modulating sine waves with other sine waves, introducing new harmonics on either side of the modulated sine wave. These harmonics would change depending on the frequency of the modulating signal, producing rich sounds with only two sine waves. Yamaha licensed Chowning's discovery and created the first FM synthesizer, the DX7, which became very popular.
In FM synthesis, the instantaneous frequency of an oscillator is altered according to the amplitude of a modulating signal. This process can create both harmonic and inharmonic sounds. To synthesize harmonic sounds, the modulating signal must have a harmonic relationship with the original carrier signal. As the amount of frequency modulation increases, the sound becomes more complex. Inharmonic sounds, such as bell-like and percussive spectra, can be created by using modulators with frequencies that are non-integer multiples of the carrier signal.
FM synthesis can be implemented using analog or digital oscillators, with digital synthesis offering greater stability. It has been used in various applications, including musical instruments, mobile phone ringtones, and sound cards for computer systems. The degree of complexity in FM synthesis can vary from simple 2-operator FM to highly flexible 6-operator engines, and it has been a key feature in synthesizers by Yamaha, Korg, and Clavia, among others.
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Waveform synthesis
Wavetable synthesis is a sound synthesis technique used to create quasi-periodic waveforms often used in the production of musical tones or notes. It was invented by Max Mathews in 1958 as part of MUSIC II, a system with four-voice polyphony capable of generating sixteen wave shapes through the introduction of a wavetable oscillator.
Wavetable synthesis allows for the optional evolution of a waveform, differentiating it from sampled synthesis, which uses static digital samples. The length of waveforms or samples may vary from a single cycle up to several minutes. The sound produced can be harmonically changed during playback by moving to another point in the wavetable, resulting in sounds that can imitate acoustic instruments or be entirely abstract. This technique is especially useful for evolving synth pads, where the sound changes slowly over time.
Digital interpolation between adjacent waveforms in wavetable synthesis allows for dynamic and smooth changes in the timbre of the tone produced. The direction of the wavetable sweep can be controlled in multiple ways, such as by using an LFO, envelope, pressure, or velocity. Additionally, wavetables can simulate methods used by analog synthesizers, like pulse-width modulation, by utilizing square waves with different duty cycles.
Generated noise waveforms, such as those used in quasi-periodic synth sounds, can be manipulated to have a perceived pitch and can be combined with other elements like envelopes to create useful sounds, particularly in percussion. Wavetable synthesis has been widely adopted by synthesizer manufacturers like PPG, Waldorf Music, Ensoniq, and Access, and is now available in hardware and software synthesizers, including apps for PCs and tablets.
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Sampling rate
The sampling rate, or sampling frequency, defines the number of samples or snapshots taken per second from a continuous signal to make a discrete or digital signal. The higher the sampling rate, the more snapshots are taken and the more complete the image. For example, at a sampling rate of 44.1 kHz, a sample is taken every 22.7 microseconds (one thousandth of a millisecond). This is considered CD quality.
The Nyquist-Shannon sampling theorem (Nyquist principle) states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled. For example, if an audio signal has an upper limit of 20,000 Hz (the approximate upper limit of human hearing), a sampling frequency greater than 40,000 Hz (40 kHz) will avoid aliasing and allow theoretically perfect reconstruction.
In digital audio, an analog-to-digital converter (ADC) takes the analog waveform and chops it up into thousands of samples per second, with each sample having one of 65,356 possible values. The sample rate is the number of samples taken per second, and this rate is important because it determines the highest frequency that can be captured and reproduced. The relationship between them is simple: to capture a given frequency, the sample rate needs to be double that frequency. So, if the upper limit of human hearing is considered to be 20 kHz, then a sample rate of at least 40 kHz is needed.
A higher sample rate also provides more flexibility when it comes to pitch-shifting and time-stretching, as it gives more room to play around before the sound deteriorates. For most music production, 44.1 kHz at 24-bit is standard, while 48 kHz is standard for video editing and production.
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Sound hardware
One of the key components of sound hardware is the sound card, also known as an audio card. It is an internal expansion card that facilitates the input and output of audio signals to and from a computer, under the control of computer programs. Sound cards can be integrated into the motherboard or used as external audio interfaces for professional audio applications. They provide flexible audio accelerator capabilities, supporting higher levels of polyphony and features such as hardware acceleration of 3D sound, positional audio, and real-time DSP effects.
The evolution of sound cards has played a significant role in enhancing the PC gaming and multimedia experience. In the late 1980s, a panel of computer game CEOs highlighted the PC's limited sound capabilities, calling for improved sound hardware. This led to the development and adoption of sound cards like the AdLib, IBM Music Feature, and Roland MT-32, which offered better audio quality and capabilities.
Another critical component in sound hardware is the Digital to Analog Converter (DAC). It converts digital audio signals into electrical signals that can be understood by speakers to generate sound. This process involves taking digital samples, which represent the audio waveforms, and converting them into the right electrical form for the speaker to reproduce as sound.
Overall, sound hardware provides the foundation for audio functionality on computers, enabling various applications such as music composition, audio editing, multimedia entertainment, and more.
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Frequently asked questions
Sound is how we, as humans, experience vibration through our ears and the rest of our bodies. We feel these patterns, or waves, of high and low pressure that cause our eardrums to move in and out repeatedly.
In order to generate sound, a computer sends electric pulses to a speaker which then creates sound. By changing the time in between these pulses, higher and lower tones are generated. Computers use digital-analog converters (DAC) to convert samples of digital approximations of physical sound into signals that drive the speakers.
Sound is recorded through "digitization" or "quantization". The height of certain waveforms is measured at certain points in time, and these measurements are stored as binary information.
