The Science Behind Panning: Ms Processing Sound Explained

The Processing Sound library allows users to manipulate sound files by panning them in a stereo panorama. This is one of many features of the library, which also includes the ability to play, analyze, and synthesize sound. The library is compatible with Processing 3.0+ and comes with example sketches to help users get started.

Volume/amplitude alteration

The volume of a sound signal can be altered through a variety of methods, each with its own advantages and disadvantages. One common method is through the use of a pan pot or panoramic potentiometer, which adjusts the volume of a signal by changing the relative amplitude between two output speakers. When the pan pot is set to the central position, the input signal is split evenly between the two output channels, resulting in a "phantom centre" image. Rotating the control to one side increases the volume on that side while decreasing it on the opposite side, creating a stereo image effect.

Another technique for volume alteration is Mid-Side (MS) decomposition, which separates a stereo signal into a two-channel signal composed of "mid" and "sides" information. The "mid" channel is created by taking the sum of the left and right channels, while the "sides" channel is created by taking the difference between the two. This allows for independent processing of each channel, such as adding equalization or compression. However, it's important to note that Mid-Side processing does not provide perfect separation between the centre and sides, and improper use can lead to undesirable effects.

In Windows, users may experience automatic volume reduction when running certain programs, such as Skype or games with audio chat channels. This is due to a feature that attempts to detect and facilitate VoIP calls by lowering the volume of other apps. While well-intentioned, this feature can be frustrating as it interferes with users' preferred volume configurations. The issue can be resolved by disabling the "Communications Activity" feature in the Sound configuration window.

Additionally, volume fluctuations can be caused by software conflicts, defective audio drivers, or enabled enhancements. To troubleshoot these issues, it is recommended to remove conflicting applications, reinstall audio drivers, disable enhancements, and perform a clean boot to identify any background programs causing interference.

It is also important to be cautious when applying Mid-Side processing techniques, as they can have unintended consequences on the overall mix. Overdoing boosts to the S (sides) channel, for example, can distort the stereo field and destroy the balance between mix elements. Similarly, applying Mid-Side compression to an entire stereo mix can result in different treatments for the same element, leading to an unnatural and unbalanced sound.

Stereo panning

Panning can be adjusted using a pan pot (short for "panoramic potentiometer"), which is a knob or slider that controls the volume of the signal in the left and right channels. By altering the volume levels in each channel, the sound can be perceived as coming from different directions. For example, a sound playing at the same volume in both left and right speakers will be perceived as coming from the centre.

The placement of instruments in the stereo field is an important consideration in panning. Typically, kick, snare, bass, and vocal tracks are panned directly to the centre as they form the foundation of the mix. Lower-frequency instruments, such as kick drums and bass guitars, are usually panned closer to the centre, while higher-frequency instruments like guitars and keyboards are panned to the sides. This creates a balanced and spacious sound.

One popular panning technique is LCR (Left, Centre, Right) panning, where tracks are panned only to the left, centre, or right channels, avoiding intermediate positions. This technique provides a wider soundstage and more space between instruments. Modified LCR panning involves panning the most important tracks to the centre and sides, while less prominent instruments are panned across the stereo field to create a sense of separation.

When panning instruments, it is also important to consider the perspective. The performer's perspective pans instruments based on their placement on stage during a live performance, while the audience perspective pans them based on their position from the crowd's point of view.

Panning is not just limited to the left-right dimension but also includes depth and height. The depth is controlled by volume levels, reverb, and time-based effects, creating the perception of distance between the listener and the "back wall" of the mix. The height is determined by the frequency range, with bass frequencies perceived as closer to the floor and high-pitched frequencies appearing higher.

Overall, stereo panning is a versatile tool that allows producers and audio engineers to create a three-dimensional soundscape, enhancing the listener's experience and ensuring that each element of the mix is distinct and audible.

Multi-channel device sound control

When dealing with multi-channel audio, it is essential to have an audio interface with a high channel count. This enables the simultaneous recording of multiple channels, each of which can be individually manipulated. For example, the Tascam US2000 audio interface offers eight microphone preamps, six dedicated line inputs, and an S/PDIF input, allowing for a high number of channels to be captured and mixed within a computer.

In terms of output control, multi-channel setups can be more complex. In some cases, the default volume control settings may be "greyed out" or locked at maximum intensity, particularly when using multi-output devices. To address this, individual volume adjustments for each device within the aggregate or multi-output setup can be made manually. Additionally, third-party applications, such as Soundflower, Loopback, or SoundControl, can be utilised to gain more flexibility in volume control for each audio device.

For those seeking a hardware solution, multi-speaker selector devices with volume control are available. These devices allow users to connect and control multiple speakers or audio zones independently, providing another layer of control in multi-channel audio setups.

Ultimately, the specific approach to controlling multi-channel device sound will depend on the user's hardware, software, and specific requirements.

Sound synthesis

Synthesizers produce sounds by generating and combining electrical signals. These signals are known as waveforms and come in different shapes, such as sine waves, square waves, and sawtooth waves. The frequency of a wave, measured in hertz (Hz), determines the pitch we hear.

There are two main types of synthesizers: analog and digital. Analog synthesizers generate sound directly from electricity by controlling electrical voltages and the route the signal takes through the synth's components. Digital synthesizers, on the other hand, operate like computers, constructing sound waves from exact values in binary code. They can be further categorized into hardware and software synthesizers.

Analog synthesizers are always hardware as they make sound by passing electricity through physical components. This results in rich, warm tones that many producers favour. Digital synthesizers, on the other hand, can be either hardware or software. Software synthesizers, also known as soft-synths, are commonly used within a Digital Audio Workstation (DAW) and tend to be more adaptable and affordable.

Synthesizers can be further categorized into monophonic and polyphonic synths. Monophonic synthesizers can only play one note at a time, making them ideal for bass or lead lines, while polyphonic synthesizers allow for playing chords.

The process of sound synthesis can be achieved through various methods, including subtractive synthesis, additive synthesis, frequency modulation (FM) synthesis, wavetable synthesis, modular synthesis, granular synthesis, and physical modelling synthesis. Subtractive synthesis involves reducing portions of the full frequency range using filters and envelopes, while additive synthesis combines partials or multiple sine waves to create complex sounds. FM synthesis uses two oscillators to generate an original frequency and introduce additional harmonics, resulting in glassy bells and electric keys. Wavetable synthesis starts with a sample of a single hit and can generate wildly different sounds from this initial shape. Modular synthesis involves deconstructing the synth into separate modules, allowing for customizable signal chains. Granular synthesis splits sound samples into very short segments called grains, which can be modulated and layered to create unique atmospheres. Finally, physical modelling synthesis simulates the way an instrument is played, such as with a mallet or a bow, to capture the nuances of a live performance.

In conclusion, sound synthesis is a versatile and powerful tool for generating sounds, with applications in music, academics, and art. By understanding the different types of synthesizers and synthesis methods, musicians and sound designers can create a wide range of sounds, from emulating acoustic instruments to exploring otherworldly realms.

Audio input analysis

The Processing Sound library offers a simple way to work with audio, allowing users to play, analyse and synthesise sound. It provides oscillators for basic waveforms, noise generators, and effects and filters to play and alter sound files and other generated sounds. The syntax is minimal, making it easy to patch one sound object into another.

The library also includes a range of example sketches to help users get started with the software. These cover many use cases and provide a useful starting point for beginners. The source code is available on the processing-sound GitHub repository, where bugs can also be reported.

The Processing Sound library has a number of functions that allow for audio input analysis. The 'AudioIn' function, for example, allows users to grab audio input from their sound card. The 'getEnergy'() function can be used to find spikes in the energy of an audio signal, which can be used to trigger a response. The 'getBeat'() function returns whether or not the current moment of audio contains a beat. The 'setBeatSensitivity'() function can be used to adjust the sensitivity of the beat detector.

The 'analyze'() function calculates the current frequency spectrum of the audio input signal, and the 'calculateFFT'() function calculates the frequency spectrum of a given audio sample. The 'getFrequency' function detects the fundamental frequency of a sound signal, and the 'getSpectrum' function inspects the underlying soundwave of an audio signal. The 'getBuffer' function gets the content of the current audio buffer from the input source.

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