Recording is actually about dealing with invisible energies such as sound waves and electrical signals. For example, a microphone converts sound waves into signals. A signal is a changing voltage that carries information. Here, it refers to the message of music.
This chapter will discuss some of the characteristics of sound and audio signals that will help you work with room acoustics, and give you an idea of what you should be doing at the mixing console when adjusting your sound. Apply this expertise to produce better recordings.
3.1 Sound Wave Creation
The vibrations of most musical instruments strike air molecules to produce sound, causing the vibrating air molecules to propagate outward in the form of sound waves. When these vibrations reach the human ear, the person hears sound.
To illustrate the process of sound wave generation, consider the vibration of a speaker cone in a guitar amplifier. As the paper cone moves outward, it squeezes adjacent air molecules together, a process called compression. The paper cone moves inward, pulling the air molecules away from each other and becoming a sparse state. As shown in the figure, when compressing, the air pressure is higher than normal atmospheric pressure, and when rarefying, it is lower than normal air pressure.
The transmission of vibrations from one molecule to the next is like the motion of a spring – the reciprocating vibrations of each molecule propagate in the form of a wave. The speed of sound waves propagating outward from the sound source is 344m/s, which is the speed of sound in the air at normal temperature.
At a certain receiving point, such as an ear or a phone, the change of the receiving air pressure is represented by the waveform movement in the figure below to represent the change of sound pressure with time. The highest point on the graph is called the crest, the lowest point is called the trough, and the horizontal center line represents normal atmospheric pressure.
The propagation of sound waves spreads out from the sound source, and the movement of compressed and sparse air molecules extends outward in a spherical shape. When a spherical wave expands, the sound pressure is distributed over a large area, and the sound pressure weakens as the distance from the sound source increases. This means that the further away from the sound source, the softer the sound. Especially when the distance from the sound source is doubled, the sound pressure will be reduced by half of the original sound pressure (a drop of 6dB). This phenomenon is called the inverse square law.
3.2 Characteristics of Sound Waves
A continuous waveform is drawn. A complete vibration process, that is, the process in which the air pressure moves from normal movement to a wave peak or trough and then returns to the starting point, is called a week. The time taken to intercept a complete cycle – the time required from the crest of one wave to the crest of the next wave – is called the period of a sound wave. A week is the length of a cycle.
3.2.1 Amplitude
The height of a waveform is its amplitude. Strong sounds have high amplitude (large pressure changes), and weak sounds have low amplitude (small pressure changes).
3.2.2 Frequency
The sound source (for example, the speaker of a guitar amplifier) vibrates back and forth many times in one second. The number of cycles of vibration completed in one second is called frequency. The faster the speaker vibrates, the higher the frequency of the sound. Frequency is measured in Hertz (Hz), which is the number of cycles per second. 1,000 Hz is called 1,000 Hz, or 1kHz for short.
The higher the frequency, the higher the perceived pitch of the sound. Low-frequency sounds have very low pitches (for example, the low E frequency on the bass is 41Hz), and high-frequency sounds have very high pitches (for example, the sounds 4 octaves above the alto C key, whose frequency is is 4186Hz).
Children can hear sounds with a frequency of 20Hz~20kHz, and most adults can hear sounds with a frequency of 15kHz or higher. Each musical instrument produces sounds in a certain frequency range. For example, the frequency range of a double bass is 41Hz~9kHz, while the frequency range of a violin is 196Hz~15kHz.
3.2.3 Wavelength
As a sound wave travels through air, the physical distance from one crest (compression) point of the sound wave to the next crest point is called the wavelength (Figure 3.1). Low-pitched sounds have longer wavelengths, and high-pitched sounds have shorter wavelengths. The wavelength is equal to the speed of dry sound divided by the frequency. Therefore, the wavelength of a sound wave with a frequency of 1000Hz is equal to 0.344m; the wavelength of a sound wave with a frequency of 100Hz is equal to 3.44m; and the wavelength of a sound wave with a frequency of 10kHz is equal to 3.45cm.
3.2.4 Phase and Phase Shift
Within the cycle of a waveform – the starting point, peak, trough or any point in between, that is, the phase of any point on the waveform is expressed in degrees, and a complete cycle is expressed in 360°. The starting point of the wave is 0°, the crest point is 90° (1/4 cycle), and the end point is 360°. Figure 3 plots the phase at each point on the waveform. If two identical waves propagate outward from the same starting point, but one wave is delayed by some time than the other, there will be a phase shift between the two waves. The longer the delay, the greater the phase shift. Phase shift is also measured in degrees, and the diagram shows that the waveform represented by the dashed line lags 90° of phase shift compared to the waveform represented by the solid line.
If you mix two identical sound waves, for example a sound wave with a reflected wave that bounces off a wall, the crests of the two waves will be superimposed at certain points in the room. The sound pressure/amplitude will then double, creating louder areas at certain frequencies.
3.2.5 Phase Interference
When the phase shift between two identical waveforms is 180°, the crest of one wave coincides with the trough of the other wave (Figure 3.6). If these two waves are combined, the waveform disappears, a phenomenon called phase cancellation or phase interference.
Suppose there is a signal with a wide frequency range, which is delayed and then mixed with the original undelayed signal. Then some frequency components will disappear due to 180° phase cancellation. This results in a hollow sound that has some timbre filtered out.
Give an example of how this phenomenon occurs. For example, when recording a singer and a guitarist, use one microphone close to the singer and another microphone close to the guitar, both picking up the singer’s voice. The singer’s microphone is close to the singer’s mouth, so there is no delay in the singing voice that can be heard from the signal. However, the guitar microphone is far away from the singer, so the singing signal it picks up is delayed. When the signals from the two microphones are mixed, we can often hear an acoustically colored sound quality due to the phase interference between the two microphones.
Suppose a stage play is recorded with a microphone mounted on a short stand on the stage floor. This microphone picks up both the direct sound from the actor and the delayed reflected sound that bounces back from the floor. When direct sound and delayed reflected sound mix at the microphone, phase cancellation occurs. When an actor walks and talks on the stage, you will hear a changed sound that is hollow and has some timbre filtered out.
3.2.6 Harmomcs
The type of wave shown in the diagram is called a sine wave. It is a pure tone of a single frequency, like the sound emitted from a sound oscillator. In contrast, most music has complex waveforms that are composed of sine waves of varying frequencies and amplitudes. A composite waveform composed of three sine waves of different frequencies is drawn.
The lowest frequency in the composite waveform is called the fundamental frequency, which determines the pitch of the sound; the higher frequency components in the composite waveform are called overtones or harmonics. If the overtone frequency is a multiple of the fundamental frequency, then these overtones are called harmonics. For example, when the fundamental frequency is 200Hz, the second harmonic is 400Hz and the third harmonic is 600Hz.
Harmonics and their amplitudes help identify the timbre or timbre of a sound, thereby identifying whether the sound is drums, piano, electronic keyboard, human voice, etc. Generally speaking, some instruments with few or weak harmonics – such as flutes – can make the sound pure and smooth, and instruments with many or strong harmonics – such as trumpets or sound-deforming guitars – can make the sound purer and smoother. The sound tends to be bright and sharp.
We will explain in detail in Chapter 10 that the timbre balance of the recorded instrument sound can be changed by improving or attenuating the balance of the wave and fundamental frequency components in the instrument sound. Boosting the fundamental component can make the sound warmer, while attenuating the fundamental component can make the sound thinner. Boosting the harmonic components can make the sound bright, precise, or have rich treble, while attenuating the harmonic components can make the sound thinner. be dimmed or suppressed.
Usually playing an instrument forcibly increases the harmonic content of the instrument, so the sound produced when the piano is played forcibly is brighter than when played softly.
Noise (such as tape hiss) covers a wide frequency range and is an irregular, non-repeating waveform. Something like a lens, a snare drum, or a singer’s “$” sound all have a hiss or noise characteristic.
3.2.7 Envelope
Another characteristic that identifies a sound is its envelope. When a note sounds, as long as it is not continuous, but rises in volume, maintains it for a short period of time, and then returns to silence, this rise and fall in volume of a note is called the envelope of the note. The envelope connects the crests of a continuous wave, thus forming a musical note. Each instrument has its own unique envelope.
Most envelopes are made up of 4 parts: attack, decay, sustain and recovery (Figure 3.8). During the build-up period, the note rises from silence to maximum volume, and then decays from maximum volume to some moderate level. This medium volume is the sustained part. During recovery, the note drops back from duration volume to silence.
For example, the impact sound when drumming, because the time of hitting the drum surface is short, the sound builds up and decays very quickly. Others, such as electronic keyboards or violins, have longer sound durations. They have a slower sound build-up and a longer duration. The sound of guitar strumming and the sound of lens impact have a faster sound build-up and a slow recovery time, so the impact sound is strong and the fade-out sound is slow.
You can shorten the decay period or ringing of guitar strings by damping the strings with the sides of your hands. Likewise, using felt attached to the roughness of a bass drum can dampen the decay of the bass drum sound, resulting in a tighter sound.
Harmonic content often changes during the envelope of a note. If an instrument has a percussive build-up period – like a guitar pick or a tom – then the harmonic content is strongest during the build-up and becomes weaker during the decay. Weaker.
3.3 Behavior of Sound in Rooms
Most music is recorded indoors, so it’s important to understand how room surfaces affect sound.
3.3.1 Echoes
The sound waves emitted by the instrument travel in all directions. Some sound waves reach the human ear directly (or reach the microphone). This sound wave is called direct sound. There are also some sound waves that will hit the walls, ceiling, floor, equipment and furniture of the recording studio. On the surfaces of these objects, some sound waves are absorbed, some are re-emitted through the surface, and some are reflected back into the room.
Since sound waves take time to propagate (sound waves move approximately 30.48cm per millisecond), the reflected sound arrives after the direct sound. There is a short delay for the reflected sound that repeats the original sound. If the reflected sound is delayed by about SOms or more, we call this reflected sound an echo (Figure 3.9). In some concert halls we can hear a single echo; in small rooms we can often hear A short, rapid succession of echoes is called a flutter echo. We can identify it by clapping near the wall. When sound bounces back and forth between two parallel walls, it creates a flutter echo.
3.3.2 Reverberation
Sound is reflected many times from all surfaces in the room. These reflections will sustain the sound of each note played by the player. The sound that remains after the original sound in the room has stopped is called reverb. The sound you hear after you yell in an empty stadium is reverberation. Your shouting in the room will linger for a while and then fade away (attenuate).
Reverb is hundreds of echoes that fade into silence. The echoes follow each other and quickly coalesce into one continuous sound. Eventually, these echoes are completely absorbed by the room’s surfaces. The timing of the echoes is random, and the number of echoes increases as the echoes decay. Figure 3.10 illustrates how reverberation occurs in a recording room.
Chaos is a continuous fade-out sound (HELL0-0-0-0-0), while echo is a discontinuous repetition of a sound (HELLOhello hello hello).
Reverberation time (RT60) refers to the time required for reverberation to decay by 60d8. Too long a reverberation time will make the recorded sound feel distant, muddy and listless, which explains why pop music is recorded in a “completely dead” or reverberation-free recording studio. This kind of recording studio The reverberation time RT60 is about 0.4s or less. In contrast, classical music should be recorded within a “live”, reverberant musical environment (RT60 is 1 – 3s). This is because we need Hear the reverberation of classical music – it’s part of the sound of classical music.
Reverberation comes from all directions. It is a mixture of sound reflections from walls, ceilings, and floors. Since we know which direction the sound is coming from, we can tell the difference between direct sound coming from the location of the instrument and reverberant sound coming from any other direction. So we can ignore the reverb and focus on listening to the sound source. In fact, our understanding of reverberation is usually incomplete.
Suppose you place a microphone next to your ear, record an instrument in a reverberant room, and then play back the recorded sound. You will find that the reverberation you hear is far greater than the sound you hear in the scene. Why is this so? This is because the reverb you’re recording isn’t just surrounding the reverb you hear, but the microphone picks up all the sounds in front of the two speakers, so it sounds more reverberant. You can’t underestimate the reverberation that exists in thousands of spaces. To reduce the amount of reverberation in a recording, you can place the microphone closer to the instrument, or perhaps add some sound-absorbing material to the room.
3.3.3 Diffusion
When sound waves hit and reflect off an uneven or spherical surface, they are dispersed or amplified. This kind of Diffusion is often used to reduce sound reflections. Sound waves can also be diffused as they propagate through small open slots.
3.4 HowtoTameEdlOeSandReverb
Echo and reverb can make recorded sounds blurry and appear distant. There are two ways to prevent these problems: using recording techniques and applying acoustic treatments.
3.4.1 Controlling Room Problems with Reoording Techniques
By following the suggestions below, you may be able to record clear sound in an ordinary room such as a club, home, or basement.
- 1.Close placement of microphones. Place the microphone 2.54~15cm close to the instrument or singer. This allows the microphone to pick up more musical instruments or vocals and less room reflections. You can also use microphones to simply attach to the instrument for pickup.
- 2.Use directional microphones—cardioid, supercardioid, or supercardioid microphones to suppress room reflections.For miking bass guitars and synthesizers, use guitar cables or D.I boxes directly. Youqian omitted the words, so the room sound will not be picked up. When miking an electric guitar, to get better sound, you should turn off the effects box or use a guitar amplifier simulator.
- See also Chapter 7 for points on reducing leakage sound.
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