Frequency (Pitch)

We all know that certain sounds are “high,” like a kettle whistling, or “low,” like the rumble of thunder. These properties of sound are commonly referred to as its pitch, and are caused by the wavelength, or the frequency, of the sound waves. Wavelength and frequency are two ways of measuring the same basic phenomenon: the length of one cycle, from peak to peak. A wavelength is plotted from one highest pressure point to the next highest pressure point. The number of these waves that pass a fixed point over the course of one second is the measure of the frequency of the sound wave. This measure of cycles per second is referred to as Hertz (Hz) and is measured along the graph’s x-axis. Sound waves travel in fairly consistent wave cycles, meaning that the pitch doesn’t change much even as sounds get quieter over distance.

A sound that generates 10,000 wave cycles every second has a frequency of 10,000 Hz, also written as 10 kiloHz, or 10 kHz. The fewer the cycles per second, the lower the pitch of a sound; the more the cycles per second, the higher the pitch (Figure 13.1).

Neither the human ear nor a microphone can perceive all sound frequencies. The range of detectable pitches for a given apparatus or organ is called its frequency range. An average, healthy human ear can distinguish pitches from 25 Hz to 20 kHz. Dogs can hear frequencies beyond 20 kHz (this is why they can hear high-pitched dog whistles that humans cannot). The frequency range that a microphone or a sound recorder can duplicate in a useful way is a common measure of equipment quality, and is called its frequency response. In terms of microphones, a typical “mic” used by a news reporter in the field would have a much more limited frequency response than one used to record an orchestra.

Amplitude (Loudness)

Each peak high and low pressure point has a specific height, or amplitude, which is a measure of the loudness of a sound and is measured on the graph’s y-axis (Figure 13.1). The higher the amplitude peak, the greater the displacement pressure of the sound wave, and the louder the sound. Loudness is measured in decibels (dB). Because the human ear can register a vast range of loudness levels, the decibel scale is a logarithmic one. We won’t go into the complexities of logarithms here, but basically this means that a small adjustment in dB can be reflected in a huge change in loudness. Another way of looking at it is that it takes an increase of 3 dB to double the loudness of a sound, and a decrease of 3 dB to halve loudness of a sound. Note that the actual decibel measurement is a pressure measurement, but the result, a measure of what we hear, is subjective.

The loudness range that the human ear can distinguish falls between the threshold of hearing (0 dB) on the lower end and the threshold of pain (120 dB) on the upper end. A normal conversational tone is approximately 55 dB. A whisper is around 25 dB and a scream is around 75 dB. At 150 dB, eardrums will rupture. In most recording situations, the loudness of your source fluctuates. Sometimes the range between the quietest and loudest sounds is minor, while at other times it can be extreme. For example, listen to the opening of Richard Strauss’ symphonic tone poem Also Sprach Zarathustra, Op. 30 (which was used in Kubrick’s 2001: A Space Odyssey). The piece begins with the softest, barely audible drone of the double basses and builds to an all-out, full orchestra fortissimo—led by crashing cymbals, blaring horns, and pounding tympani—in only a minute and a half! The range of different loudness levels in a scene, or musical sequence, is referred to as its dynamic range. Also Sprach Zarathustra has an extremely wide dynamic range. Comparatively, a song like the White Stripes’ Fell in Love with a Girl has a narrow dynamic range because it remains at the same loudness level throughout. A conversation that goes from a whisper to screaming has a wide dynamic range, whereas a politician’s speech delivered in a monotone has a narrow dynamic range. Wide dynamic ranges can be challenging for both the sound recordist and the recording equipment.

■ Figure 13.2 Waveforms for a piano playing middle C (left), a violin playing middle C (center), and a human voice singing middle C (right).

Quality (Timbre)

The sound waves shown in Figure 13.1 represent pure, electronically generated tones with no character or aberrations. The waves of sounds from the real world are not quite so uniform, as they lack a smooth curve and perfectly symmetrical peaks and dips. Most naturally occurring sound waves include characteristic irregularities in the overall shape, which dictate the particular quality of that sound.

The central and dominant shape of the wave is called the fundamental tone, but every fundamental tone also resonates with a series of imperfections and coinciding waves that are known as overtones and harmonics. These elements constitute the timbre of a sound, its unique tonal composition and characteristics of that sound (its richness, harshness, or resonance, for example). Timbre allows us to easily distinguish different instruments playing the same note. For example, middle C on a piano sounds quite different from middle C played on a trumpet, or on a guitar, or when sung by a human voice (Figure 13.2).

Velocity

As a wave that travels through air, sound has both directionality and speed. At a temperature of 60 degrees, the speed of sound is 1,117 feet per second (about 762 mph). This is very slow compared to the speed of light (which is 983,571,056 feet per second). This is why, when you’re watching a fireworks display, you see the big flash of light first and hear the boom of the explosion seconds later.

Sync Sound

Sync sound is recorded with the image, so sound and picture correspond with frame accuracy, and are said to be in sync (Figure 13.3). Sync sound could be an interview, a conversation between people at a dinner table, or the sound of a door closing—anything in which the sound emanating from the scene is recorded simultaneously with the picture.

Sync sound can be recorded using a single-system or a double-system process. In single-system, audio and video are gathered at the same time, with the same apparatus (the camcorder), and are recorded in sync on the same media (videotape or memory card). In double-system recording, sound is recorded on a separate audio recorder and combined with the picture in postproduction.

■ Figure 13.3 The documentary sound crew often consists of one person who is in charge of recording sync sound. Here, sound recordist JT Takagi is working with Director of Photography Michael Chin and Director Megan Mylan on the short documentary My Little Friends (2013). Photo courtesy of Principe Productions/PrincipeProductions. com.

Most digital video cameras have the capability to record both audio and video, ensuring that sound and picture are automatically and always in sync. Film, on the other hand, is a double-system sound medium, meaning the film camera records the image, and a separate sound recorder gathers the audio. Historically, documentarians working in video used single-system, and the sound recordist would send the audio signal to the camera via a breakaway cable (Figure 13.4, left). Double-system is now used frequently with DSLR shooting (Figure 13.4, right) because of the audio limitations of many DSLR cameras. DSLRs, created for the consumer market, are equipped with miniplug connectors for microphone inputs. These are notoriously unreliable, as well as unbalanced (and therefore more prone to interference), and are replaced on more professional cameras with the sturdier XLR inputs.

■ Figure 13.4 In single-system recording (left), cables connect the audio and video equipment. With double system sound (right), the audio recorder is separate from the camera. Because the sound recorder and camera work independently, a slate is used so that picture and sound can be synced in postproduction.

In documentary practice today, the line between single- and double-system sound is further blurred by the use of wireless technology. To avoid having the camera and sound recordist connected by an unwieldy cable, many sound recordists send the signal from their portable field mixer or recorder (p. 219) to the camera using wireless technology. This is single-system sound, but because they will usually run a backup digital audio recorder as well, it can be considered a hybrid setup (Figure 13.5).

■ Figure 13.5 Audio signal path for various documentary sync sound recording setups.

Double-system sound always requires the additional step of syncing audio to the picture in post-production. This has traditionally been done through the use of a slate. The slate is used to create a one-frame, easily identifiable reference “moment” with which to line up the picture and sound. That moment is the sharp closing of the slate, which is recorded by the camera and the audio recorder at the beginning of every take. Later, in your nonlinear editing system (NLE), it is easy to find the exact frame where the slate closes. Then, you simply line up the clap image with the corresponding audio and everything after that point, for that take, should be in sync (Figure 13.6). There is more on syncing picture and sound in Chapter 17.

Many professional audio recordists prefer to work double-system because, as mentioned earlier, they are freed from having to be connected with the camera by a cable. Many work with time code-equipped digital audio recorders, like the Sound Devices 633 field mixer/multi-track digital audio recorder. By syncing (or jamming) the time code of their recorders with the time code on the camera at the beginning of the shoot, the picture and sound can be easily synchronized in the editing room.

■ Figure 13.6 A properly shot and recorded slate is the perfect audiovisual point of reference in postproduction. The frame where the slate is clapped is marked in the video file (left), and then in the sound file (A). The clips can then be merged into a new clip with video and audio in sync.

For students and others without budgets for expensive recorders and wireless transmission systems, shooting single-system is still safer and quite common. If at all possible, use a breakaway cable that sends two channels of audio to the camera and then brings the output of the camera back to the sound person so they can monitor the recorded signal through headphones (Figure 13.7).

There are advantages and disadvantages to shooting single- or double-system sound. With single-system, the difficulties of slating and syncing the audio with the picture in postproduction are avoided. On the other hand, single-system requires that the sound recordist be physically connected with the cameraperson via a cable (or use an expensive wireless transmission system). In addition, single-system means that you need to be mindful of the input levels on the camera itself. Unless you are working with a portable field mixer (pp. 219–220), this places an additional burden on the cameraperson, whose main job is to be concentrating on the image (not the audio levels).

■ Figure 13.7 A breakaway cable allows two channels of audio to travel from a field mixer into the camera, and brings an audio signal back so the sound recordist can monitor the output of the camera. A “quick release” connector allows the sound recordist to separate quickly from the camera operator.

Double-system recording allows the sound person more mobility in finding the best microphone placement, and to set their own recording levels (Chapter 14). Syncing sound in postproduction is also becoming easier because of software programs, like Singular Software’s PluralEyes, that allow for rapid and fairly mechanical syncing of sound and picture.

in practice

■ Changing Sound Recording Setups

Daniel Brooks (Figure 13.8) is a sound recordist who has worked on more than 100 documentaries and TV programs over the past 25 years. He summarizes the history of double- and single-system recording this way:

When I started (in the 1980s) we still had Nagras with open reel magnetic tape, and you could only record 11 or 13 minutes on a 5-inch reel, and then you had to change it. And every time you started and stopped you had to sync with the camera. It was really not conducive for any sort of vérité or reality shooting because you had these very small windows. With video it became single-system, and we sound people became the “pull-toys” of the camera operators because we were cable-connected to them. Now wireless mic technology has gotten much better and we are using radio transmitters and receivers to send the sound to the camera, while recording backup in audio recorders we carry on us. So we’ve gone back to double-system, which is very exciting for me because it allows sound to not be attached to the camera, and to go where the sound is, instead of where the camera is. Now I can roll when I hear something good, and I don’t have to run over the camera person and say “Roll!” and have him say, “I’m still focusing, I’m not ready yet.” If I have the great soundbite recorded the editor can cover it with a cutaway, and build a scene around it. It’s great having the ability to be autonomous.²

■ Figure 13.8 Sound recordist Daniel Brooks carries a six-channel mixer/recorder on a harness, with wireless mics and a shotgun on a boom. Here he is shown shooting single-system sound on location for Holly Hardman’s Good People Go to Hell, Saved People Go to Heaven (2013).

Wild Sound

Wild sound is audio that is recorded on location, but not simultaneously with the picture. The most common type of wild sound recorded on a documentary shoot is room tone, which is the ambient sound of a location when nobody is talking (Chapter 14). Another type of wild sound is wild sound effects. Often when recording dialogue or interviews, sound recordists will try and create the quietest environment possible in order to get the “cleanest” recording of the dialogue. Later, they will record sounds from the environment that can be added in to help build a richer, more complex soundtrack. JT Takagi, a filmmaker and sound recordist who works with Third World Newsreel (a progressive alternative media center that trains, distributes and produces media by and about people of color and social justice issues) explains:

From time to time, you might be unable to get a specific sound because of microphone placement, or simply because you missed the opportunity to record the sound when it occurred. In these cases, a sound recordist will often rerecord specific sounds from the scene as wild sound so the editor can insert them in postproduction. For example, in a documentary interview situation, when the sound of a plane overhead or a car passing interferes with the recorded sync sound, the sound recordist will often wait until the interview is over and then ask the interviewee to repeat certain words or phrases and record them wild so they can be inserted during editing.

Sometimes wild sound can be recorded and used as a sound design element. For coauthor Kelly Anderson and Allison Lirish Dean’s film My Brooklyn (2012), about a lively commercial area in Downtown Brooklyn, the filmmakers recorded wild sound of the location—people selling cellphones, hip-hop emanating from storefronts, conversations on the street, buses passing by—that could be layered on multiple tracks in postproduction to add specificity and character to the representation of the place (see Chapter 21 for more on sound design).

Room Acoustics

Sound recorded in a loft with hardwood floors and big windows will be quite different than sound recorded in a carpeted room with window drapes. An environment with hard surfaces is called live and is undesirable for sound recording because a microphone will pick up the audio directly from the source as well as the audio bouncing off the walls and floor. The result is a boomy or echoey sound as the signal duplicates itself over and over again, creating reverberation (or reverb). The carpets and furniture in the second room, however, are poor reflective surfaces and serve to absorb sounds after they leave the source. This is known as an acoustically dead recording space. Sound recordists often use sound blankets (or comforters or rugs) to create a deader environment for recording. You can’t get rid of “boominess” or reverb in postproduction, but you can always add it to a track later on. So the goal during recording is typically to record under the deadest conditions possible.

Another factor that affects acoustics is room size. A small tiled bathroom is a very live space, but the reverberation intervals will be shorter than in a studio loft, where the sound travels a greater distance to a reflective surface and back to the microphone. What this means for sound recording is that in the bathroom, the reverb will be less problematic, though it may still be noticeable.

in practice

■ Working in Live Environments

Sound recordist Daniel Brooks (Figure 13.9) tells about a time he had to record in a very live room for a documentary about the architecture of telephone buildings:

The only place we could shoot the interview was in a big empty room that had metal floor plates and hard walls and glass windows, and it was super live.

Over the years, I’ve acquired these sound absorbing panels that you can put in your home recording studio. I found some of those in a dumpster, and I glued them onto some corrugated plastic. I brought those to the telephone building and put them up, as well as sound blankets, and it sounded great! It had been an utterly unrecordable space before, and then, by knowing what direction the person was going to be speaking in, and putting sound panels to catch the sounds before they hit the walls and bounced to the ceiling and hit the floor, I was able to make it quiet.⁴

■ Figure 13.9 Daniel Brooks’ alternative to sound blankets is a homemade sound panel made of studio foam baffle mounted on a lightweight backing.

The Basic Signal Path

Let’s look at the basic signal path of a sound in a particular digital audio recording situation. Sound starts out as an acoustic source, which is transformed into an analog electronic signal, then turned into digital data, only to be transformed back into acoustic sound again (Figure 13.10):

1. Sound recording begins with the source of sound, which emits acoustic energy (sound waves).
2. These sound waves enter the microphone, which converts the acoustic energy into fluctuations of electrical voltage that are analogous to the original sound waves. The electronic signal is analogous, or like the original sound wave, in two ways. When the sound is loud, the electronic voltage goes up. When the sound is high pitched, the electrical waves’ frequency will increase correspondingly. This fluctuating voltage created by a microphone is called a microphone (or mic) level signal, which is sent to the digital audio recorder (or a camcorder) via a microphone cable.
3. The relatively low-voltage mic signal first passes through a preamp, where the signal is boosted, and then goes to an analog-to-digital converter (ADC), which samples the analog audio information and translates it into binary code (a series of 1s and 0s). The binary data represents the characteristics (frequency, amplitude) of the source sound.
4. The digital information is stored as a file on some form of recording media (this could be a hard drive or memory card, depending on the recorder you use). In the case of camcorders, the audio is recorded onto tape or memory cards and is linked to the picture files.
5. When playing the audio back, the data is sent to a digital-to-analog converter (DAC), which changes it back into electronic energy and outputs a line level signal, which is the audio signal used between audio devices such as cameras, mixers, and amplifiers.
6. The audio line signal can then travel to speaker amplifiers or headphones, which convert the signal back into sound waves that travel through the air and are received by our ears.
7. The audio data can also be sent digitally to the hard drive of a digital editing system.

■ Figure 13.10 The basic signal path of sound in digital audio recording.

Balanced vs Unbalanced Audio Signals

An important characteristic of the sound flowing through your cables and connectors is whether it is balanced or unbalanced. A balanced audio signal runs through a three-wire cable: two wires for the signal and a hum-resistant shielding cable, which is usually grounded. The signal can travel long distances with no quality loss, and is relatively impervious to distortions like hum and RF (radio frequency) interference. A balanced signal usually travels through XLR connectors, which are found on all professional digital video equipment, including microphones.

Unbalanced audio, on the other hand, is found in consumer cameras that use a miniplug connector for audio input. These are highly unstable (because they can easily pull out) and prone to interference and hum. (See p. 216 for workarounds for cameras with miniplug connectors.)

Digital Audio Quality: Sampling Rates and Bit Depth

Regardless of your recording medium, you will have choices about what sample rate, bit depth, and even file type to use when you are recording sound. These parameters affect recording quality and are important factors in ensuring a smooth high quality workflow through postproduction. These settings define the ADC process, especially as it relates to how thoroughly and accurately the analog information is measured before it is recorded digitally.

■ Figure 13.11 The process of converting analog sound (1) to a digital format involves sampling the signal at regular intervals. The lower the sampling rate (A), the less accurate the digital version will be; a higher sampling rate (B) creates a more faithful reproduction of the original sound.

Audio sample rates determine how many times a sound is measured or “sampled” per second. One sample is a single measurement of the sound wave, like a snapshot of a piece of that sound. The more the samples (the higher the sample rate), the more accurate the reproduction will be because amplitudes will be measured more often, giving a better picture of both amplitude and frequency variations (Figure 13.11). Higher sample rates produce better quality sound, but they take up more space on your storage medium. The most common sample rate for recording audio on either a camcorder or a digital audio field recorder is 48 kHz (that is, 48,000 sample measurements per second). As a point of comparison, the standard sample rate for audio CDs is 44.1 kHz (you’ll find this sample rate on some recorders and camcorders). On high-end audio recorders, you’ll find sample rates up to 96.096 kHz or even higher. For most documentary situations, 48 kHz should be more than adequate.

Bit depth is a measure of the accuracy and detail of each audio sample, determined by the number of binary digits (bits) assigned to each sample; this is also known as the sample size. The greater the bit depth, the better your audio quality will be because the sound wave, in all of its complexity, is more accurately defined. Imagine having a ruler that is divided into 1/4-inch units. If any measurement falls between the 1/4-inch marks, it will be rounded up or down. This ruler doesn’t give you particularly accurate measurements. Now imagine a ruler that is divided into 1/48-inch units and another that is divided into 1/96-inch units. These rulers will measure far more accurately because measurements that fall between markings need to be rounded only slightly. Bit depth works the same way, with a sound being measured more or less accurately, through the number of sampling “levels.” A 4-bit sample will measure 16 possible levels, an 8-bit sample will measure 256 possible levels, and a 16-bit sample will measure 65,536 possible levels. With each bit you add, you double the number of values that can define that sound, so with 24-bit audio, there are 16,777,216 possible levels! With greater depth, a more accurate picture of the original audio wave can be rendered. For areas of the wave that are not measured directly, the equipment will round up or down in a process called quantizing. With more bits, you reduce the quantizing error of the recording. In the field, you will often encounter 12-bit audio (substandard), 16-bit audio (good quality), 20-bit audio (better quality), and 24-bit audio (superior quality generally only found on professional equipment).

The mechanism for converting an analog signal to digital data via sampling is called linear pulse code modulation (LPCM) audio. LPCM audio is an uncompressed encoding method, and it is by far the most pervasive digital recording process for professional audio field recorders and video camcorders. The most popular audio file formats for audio field recording, .WAV (PC standard format) and .AIFF (Mac standard format), use LPCM encoding. So, to summarize, the standard sample rate and bit depth settings for high-quality media production audio are 48 kHz and 16-bit, but if you have the capability and storage space to go to 48 kHz and 24-bit, then by all means use those settings.

Sound Recording on Video Camcorders

Despite the increasing popularity of double-system recording, many documentary filmmakers are recording single-system sound with their camcorders, and you can get great audio this way. Devices that use tape as their record medium write LPCM digital audio along with the picture signal. Cameras that use file-based media create picture and sound files simultaneously. While most camcorders appear on paper to have excellent audio specifications, there are mitigating factors that can present problems, particularly with lower-end equipment.

■ Figure 13.12 Although it is possible to get an XLR-to-mini cable (top) to use professional microphones with 1/8-inch microphone inputs, a mountable adaptor like the Beachtek (bottom) provides a sturdier solution.

Miniplug audio inputs are especially a problem with low-end camcorders, as well as for DSLRs. These connections are fragile and prone to poor contact, and the cables are unshielded and unbalanced. Some people use an XLR-to-mini adaptor (called a pigtail) so that they can use professional external microphones. This, of course, is better than nothing, but the problem with this solution is that it converts your lovely balanced, shielded audio into an unbalanced signal, vulnerable to interference and noise. We have had the experience of shooting with a professional microphone connected to the camera with an XLR-to-mini adaptor, only to find that AM radio was being picked up on our tracks.

Many people use camera-mountable adaptors with preamps and XLR connections for cameras that have only miniplug audio inputs (Figure 13.12). These adaptors allow you to use XLR cables, and some even provide a shielded cable to the camera. However, where you find miniplugs, you may also find cheap preamps and audio circuitry, which will add system noise to your signal.

Professional cameras with XLR connectors will have two microphone inputs with independent level controls. This is important as you will often be using two microphones to record different sync sounds at the same time (by having one mic on each person in a two-person conversation, for example). You want to make sure you have the ability to record these as separate tracks and that you can set and monitor their levels independently. In Chapter 14, we explore in detail the proper method for setting levels manually.

Controlling the record levels of your audio carefully is a key to good sound. Unfortunately, the preset for many consumer-grade cameras is automatic gain control (AGC), which automatically sets your record audio levels. As with autofocus and auto-iris, in most instances you should turn off this blunt tool and set your levels manually. The problem with AGC is that it tries to bring every sound to a middle volume, regardless of whether it is a shout or a whisper. It is constantly responding to peaks and pauses in audio, adjusting levels up when there is quiet and down when a loud noise occurs, however briefly. The background sounds, too, rise and fall very noticeably with each auto adjustment. Like any prescriptive advice, though, this recommendation can be ignored at times. If you are going as a one-man-band to a noisy demonstration, you should probably set your audio level to automatic and concentrate on your shooting.

The Digital Sound Recorder

If you are shooting with a DSLR camera, or want to record double-system sound as a backup for wireless transmission to the video camera, you will be using a digital soundrecorder to record your audio. All portable digital sound recorders used for documentary production record LPCM audio and are essentially the same in their basic features and operation (Figure 13.13). These features include microphone inputs, record level controls and meters, recording quality settings, and audio outputs. Increasingly, digital sound recorders are also mixers, capable of handling and combining up to a dozen or more sound inputs.

■ Figure 13.13 Professional sound recorders have the same features: level control potentiometers (A), a peak meter (B), controls for record, play, and searching (C), analog/digital audio inputs (D), headphone jack (E), audio outputs (F), record media bay (in this case compact flash memory (G), and digital audio inputs and outputs (H).

Level Controls and Meters

Adjusting and monitoring the strength of your audio signal is at the heart of the sound recordist’s craft. The term levels refers to the strength of your audio as it enters the recorder, and the degree to which you boost or lower that audio with manual level controls, sometimes called gain controls or pots (short for potentiometers). This adjustment determines the strength of the recorded audio signal and is called setting levels. On professional recorders, you will have one level control for every micro-phone channel, allowing you to adjust the levels of each microphone independently. Setting levels is aided by a peak reading meter (Figure 13.14). The peak meter is a highly sensitive instrument that measures and indicates the level of every sound entering the recorder. Each mic input will have its own cor responding peak meter. Meter displays can be quite different from machine to machine—they can involve pivoting needles, colored LED lights, or backlit LCD displays. Whatever indicators they use, all displays are calibrated in decibels that run from –∞ dB on the extreme low end, through –40, –30, and –20 dB, and so on, to 0 dB on the high end. At –∞ dB, there is no signal at all and you will record only system sound. If your signal strength approaches 0 dB, you are already recording at too high a level and your sound will be distorted. This can be confusing, as 0 Db would seem to refer to silence, not an optimal sound level. We will discuss the reasons for the difference, and more about setting record levels in Chapter 14.

■ Figure 13.14 Peak meters are essential for monitoring the strength of the signal being recorded. They can be LED based (top, showing a two-channel meter) or LCD based (bottom, showing a six-channel meter with two outputs at the top.)

Digital Recording Media

An important aspect of digital audio recorders is their recording media format, which refers to the media they use to store the audio data. Most audio field recorders record in the .WAV file format, but how they store these .WAV files differs. Digital recording formats come and go as technology evolves, but there are a few standard ones that you are likely to encounter.

Flash Memory Recorders

Sound recorders using non-volatile flash memory cards are a popular and relatively inexpensive choice. Almost every major sound recording equipment manufacturer has developed portable flash recorders. These recorders do not have internal hard drives. Instead, they record audio directly to data cards, typically either CF or SD cards. From these, the sound can be transferred onto computer hard drives for storage, and the cards can be reused again and again. Compact flash cards also contains no moving parts, which means they are reliable in extreme conditions (Figure 13.15).

Hard Drive Recorders

Hard drive recorders write their data directly to a hard drive. Most portable units intended for media production use a 2.5-inch solid-state drive (SSD) with a capacity of up to 250 GB or more. Depending on the size of the hard drive, these recorders can store many hours of audio.

Hard drive recorders also interface with computer editing software seamlessly and have a reputation for being quite robust because temperature, humidity, and motion have little effect on their functions and recording. The benefits of hard drive recorders come at a price, how-ever, as they are notably expensive. Ultra-high-end, professional hard drive recorders include simultaneous secondary media recording to flash memory or a tertiary storage device, like an external hard drive. The Zaxcom Zax-Max can even send a wireless signal to a camera or other external device (Figure 13.16).

■ Figure 13.15 Compact flash memory cards allow for small, yet sturdy digital recorders like this Zoom H4n.

Portable Field Mixers

Portable field mixers (also called microphone mixers) are small audio consoles that allow for independent level control of multiple microphone inputs (usually from one to four). You can combine the inputs in various ways and output two channels of sound to your camera (Figure 13.17).

■ Figure 13.16 Some high-end digital sound recorders like the Sound Devices 788T (top) store data directly to an internal solid state drive as well as to flash memory cards. The Zaxcom Zax-Max (bottom) sends a wireless signal directly to a camera or other external device.

■ Figure 13.17 Field mixers allow precise control over the recording levels of several inputs. Pictured are 2 three-channel mixers, the Sound Devices 302 (left) and the Shure FP33 (right).

Many sound recordists working with single-system set-ups find portable mixers an indispensable tool, because camcorder level controls are located right on the camera and it can be very awkward to have the sound recordist hovering around the camera setting levels. Using a field mixer enables the sound recordist to monitor and control levels at a distance from the camera. A mixer also allows the recordist to use multiple microphones and select which signals to send to the camera. The output of a field mixer can connect with the camera via XLR cables, a breakaway cable or via a wireless connection. It is vital, however, to calibrate the gain levels of the mixer and the camera so that you maintain audio level consistency. This process, called setting tone, is discussed in detail in Chapter 14. Located in the signal chain between the microphones and the camera audio input, field mixers are small enough to be worn in a carrying case over the shoulder.

While mixers and audio recorders were traditionally separate devices (Figure 13.18, top), a recent development is the integration of the mixer and the digital audio recorder, as seen in the Sound Devices 664 recorder (Figure 13.18, bottom). These allow the recordist to both control levels and record audio files. They can also send audio to the camera via cable or wireless. They can record up to 16 channels of audio and are becoming quite standard in professional sound for documentaries.

■ Figure 13.18 A modest audio recording setup involves a portable field mixer and separate audio recorder (top). Another approach is to use a combination mixer/recorder like this Sound Devices 664, which can mix six channels to two as well as record each channel separately (bottom).

Microphones

Simply put, a microphone is a device that converts acoustic energy (sound waves) into electrical energy (electrical signals). All microphones are constructed with a diaphragm, a thin membrane that is extremely sensitive to the vibration of air particles. The vibrations of the diaphragm, which correspond to the sound waves buffeting it, are translated into fluctuating voltage. One of the ways we identify different microphones is by the method they employ to make this conversion.

Dynamic, Condenser, and Electret Condenser

A common type of microphone for field production is the dynamic microphone, which generates a signal through electromagnetic principles. This microphone is sometimes called a moving coil microphone because the diaphragm is connected to a wire coil with a permanent magnetic charge. This coil is called the voice coil and is suspended around a permanently fixed magnet. As the diaphragm responds to a particular sound source, the coil moves up and down with the vibrations of the diaphragm. Each movement of the coil through the electromagnetic field that surrounds the magnet produces an electrical current that is analogous to the original acoustic vibrations (Figure 13.19, top).

■ Figure 13.19 A moving coil microphone (top) works by converting the movement of a diaphragm (A) into an electrical charge when the coil (B) attached to it moves up and down while suspended around magnets (C). A condenser microphone (bottom) uses a positively charged diaphragm (D) and a negatively charged back plate (E) to form a capacitor; the movement between these two electrically charged plates creates voltage fluctuations that are sent to a preamp (F). In order to keep the backplate charged, this mic requires a power supply (G).

Dynamic microphones are renowned for their rugged construction, which makes them a favorite for shooting in rough weather, high humidity, or around heavy machinery. They are also less expensive than other types of microphones. As a general rule, dynamic mics are faithful to the original sound, and also have a fairly good frequency response that is especially appropriate for the human voice. In close mic situations, they are more than adequate, which is why news reporters, who don’t mind having the mic in the shot, usually use these mics. When the microphone needs to be further away from the subject, or greater frequency response is necessary, recordists usually turn to the condenser (or electret condenser) microphone.

Condenser microphones use a diaphragm to translate the sound waves into electrical energy, but instead of using a magnet they use a capacitor to create the electrical signal. The capacitor, or condenser, is made of two round plates oriented parallel to each other, with a very narrow space between them called the dielectric. One plate is the microphone’s diaphragm, a movable acoustically sensitive membrane; the other is a fixed plate called the back plate. Both of these plates are charged with polarized voltage. When the plates are close, they can store a certain amount of electricity. When they are further apart, they can store less and the current drops. When sound waves move the diaphragm, the voltage relationship between the plates creates the electrical signal. Because there is no heavy magnet to move, the resulting signal is more sensitive, especially when recording higher frequencies. The output signal of this capacitor is very low, however, so condenser microphones have a preamplifier (“preamp”) built into the microphone (Figure 13.19, bottom).

In order for the microphone’s capacitor to work, both plates require some source of power to provide the necessary polarizing voltage, and the preamp also requires some power. Condenser mics can be powered through the use of phantom power, which is power provided by the camera, audio recorder or mixer, delivered to the microphone via one of the three XLR cable prongs, or through the use of a battery power source, which is usually located in an intermediary capsule connected to the microphone (Figure 13.20).

■ Figure 13.20 This Sennheiser ME66 electret microphone (top) uses a battery to provide the power necessary to charge the capacitor. The Schoeps SuperCMIT2U condenser microphone (bottom) needs to pull phantom power from a field mixer or other source.

Electret condenser mics offer a less expensive alternative. Though not quite as sensitive as condenser mics, they are still superior to dynamic ones in terms of sound quality. The electret is a piece of electrically sensitive ceramic that generates its own electricity when squeezed. This means their condenser is permanently charged and they need only a small amount of power for their preamp. This is usually provided by a small battery located in the microphone itself (AA, or the smaller N battery, or the even smaller LR44 1.5 volt). The low power requirements allow for a more compact design, which is always welcome in field production. Price-wise, electrets occupy a middle ground between condenser mics, which are expensive, and dynamic mics, which are less expensive. They are the mics most likely to be used on student or lower-budget productions.

■ Figure 13.21 The standard professional audio cable is the XLR, a tough and inexpensive solution for sending balanced, distortion-free audio between microphones and recorders (including cameras).

Most professional-quality microphones send a balanced output utilizing the standard XLR professional microphone connector (Figure 13.21). This shielded connection greatly protects the signal from interference caused by AC, fluorescent hum, or radio frequencies. The other advantage of XLR connectors is that they are rugged, and the male end of the connector fits with the female end through a tongue-and-groove fit and a spring lock, providing for a strong and stable connection that cannot be inadvertently pulled loose.

Microphone Frequency Response

Frequency response refers to the sensitivity of a given microphone to the range of high and low frequencies in the sound spectrum. This measurement is represented by a frequency response graph (Figure 13.22). The x-axis on this graph measures the micro phone’s response in dB, and the y-axis measures the frequency of the recorded sound. A perfect microphone would have an equal response throughout all frequencies of the sound spectrum, resulting in a flat line. This is known as a flat response. For all mics, however, the response dips at the extremes of their capabilities. All professional microphones come with a spec sheet that will indicate the instrument’s frequency range.

■ Figure 13.22 A frequency response graph plots how sensitive (measured in dB, on the x-axis) a microphone is to a range of frequencies (measured in Hz on the y-axis). Note that the Schoeps MK41 microphone (top) has a very flat response, meaning that it favors all frequencies equally (except extremely low ones). The Audio Technica AT803b, an omnidirectional lavalier microphone (bottom), has a “brighter” high end to emphasize voices, and a “roll-off” option that cuts out low frequencies at the arrow.

Some microphones come with a low-end roll-off switch that makes them less sensitive to low frequencies. This can be useful in situations where there is wind or traffic noise in the field. A roll-off switch usually has two symbols (Figure 13.23), and it is critical that you check your microphone before shooting to see which setting it is on. Documentary, which presents many challenges for the sound recordist, requires that you use your judgment when deciding whether to record “flat” or use the roll-off setting.

■ Figure 13.23 Some microphones have a low-end roll-off setting (B) that makes them less sensitive to low frequencies, usually caused by wind or machine noise. If you prefer to record the full range of frequencies in your environment, record “flat” (A).

Microphone Directionality

An additional way of defining microphones is by their directionality (also called pickup pattern). A microphone’s pickup pattern dictates the area and range within which the microphone will respond optimally. In simple terms, some microphones record sound all around them, while others favor the sound in a particular direction. Directionality is sometimes described as a microphone’s angle of acceptance.

Omnidirectional

An omnidirectional microphone (Figure 13.24) picks up audio from all directions equally. This microphone is a good choice for recording general ambient sounds (like crowd noises) or for recording a scene where sound emanates from a number of different directions (e.g., four friends gathered around a table for dinner). This is a good choice for interviews in which you want both the interviewer and the interviewee to be recorded equally.

■ Figure 13.24 The pickup pattern of an omnidirectional microphone allows it to capture sound equally from all directions.

■ Figure 13.25 A cardioid microphone has a pickup pattern that favors sound coming from the front and sides, but not from behind.

■ Figure 13.26 A hypercardioid microphone pickup pattern has a narrow range of acceptance and increased sensitivity. It greatly favors sound coming from the front and not from the sides or back.

Cardioid

The pickup pattern of a cardioid microphone (Figure 13.25) is just as its name suggests: heart shaped. The pickup pattern is somewhat directional, so the mic can be aimed specifically at the source of the audio. This mic minimizes extraneous noise while providing a natural ambient feel. Its sensitivity is primarily in front, with some sensitivity to the sides, but the mic picks up very little from behind, which is usually where the equipment and crew are. This is one of the most common microphones used in documentary production.

Hypercardioid and Shotguns

Hypercardioids and shotgun mics (also called supercardioid or unidirectional) duplicate the heart-shaped pickup pattern, but these mics are considerably more sensitive to sound directly in front of them (Figure 13.26). Their pickup patterns are highly directional, meaning that they are considerably narrower than a cardioid and can be held at a greater distance. A full shotgun mic is extremely sensitive and the most directional of the two. Often recordists will use the hypercardioid, which is directional but more forgiving if your placement is slightly off, and a lot shorter and hence easier to use and less imposing. There are drawbacks, however, to using both of these microphones. Because these mics are so sensitive, you must be careful when using them indoors. Not only will they pick up the sound directly from the source, but they can also easily pick up the reflections of that sound, resulting in audio with a “boomy quality.” These mics are quite successful outdoors, but here, too, you must be careful about sounds within the mic’s direct pickup pattern that might not be noticeable to your ear. For example, a camera-mounted shotgun or hypercardioid mic will pick up not only the voice of the subject it is pointed at, but also the truck that is half a block away behind them.

Microphone Usage Types

Microphones are also characterized by their form and function. Here are a few common types.

Handheld

They are commonly used by news reporters and others who don’t mind if the mic appears in the frame. Generally dynamic, they usually have an omnidirectional or cardioid pickup pattern. It’s a good idea to have one of these sturdy options in your kit as a backup.

Shotgun

A shotgun mic is a mic designed to be supported by a shock mount on a boom pole or a pistol grip (Chapter 14). A narrow pickup pattern and excellent frequency response makes these mics the workhorses of documentary field production. Both mics in Figure 13.20 are shotguns commonly used in documentary. We will explore techniques for using this mic in Chapter 14.

Lavalier

Lavalier microphones (also called “lavs”) are tiny clip-on mics that can be attached to a lapel or tie, or easily hidden under a collar (Figure 14.4). They usually have an omni directional pattern and are the typical mic solution for an interview where the subject won’t be getting up and moving around. These are electret condenser mics, and their power supply and preamps are in a capsule separate from the actual microphone head (Figure 13.27).

■ Figure 13.27 The Sony ECM-44 lavalier is a commonly used microphone for interviews where the subject isn’t moving around a lot.

Lavs are often used with wireless transmitters for observational shooting as they free the subject from either a cable connection or a boom while their position close to the subject’s mouth provides a high signal to noise ratio. They also allow the subject to be very far from the cameraperson, when booming is impractical because the boom would appear in the shot. In Ilisa Barbash and Lucien Castaing Taylor’s Sweetgrass (2009), for example, wireless lavs were used extensively to capture the voices of sheepherders at distances of up to a mile from the sound recordist and camera person (p. 349).

Wireless Microphones

Wireless microphones (also called radio mics) consist of a small pocket-sized transmitter to which a microphone (very often a lavalier) is attached. The transmitter sends the electrical audio signal via VHF or FM frequencies to a receiver that is connected to the input of the digital audio recorder or the camera. Wireless lavaliers are extremely common in documentary because they allow close miking of subjects while maintaining freedom of movement (Figure 13.28). Wireless microphones are also especially advantageous to the one-man-band shooter, who will put wireless lavaliers on one or more subjects and feed the signal directly into the camera (Chapter 14).

If there is a downside to wireless microphones, it is that they are expensive and vulnerable to interference, especially as the transmitter is moved away from the receiver. Some systems use a “diversity” mechanism that is constantly searching for the clearest transmission path. As with most things, the more expensive the microphone, the less prone to interference it is. You should always have a hard-wired solution available as backup.

■ Figure 13.28 This Lectrosonics L series hybrid digital wireless microphone system includes a transmitter (left) and a receiver (right) and features an M152 lavalier head.

Pressure Zone Microphones (PZMs)

Pressure zone microphones (PZMs), also known as boundary mics, are specialized mics mounted on a plate, typically metal (Figure 13.29). One of the main advantages of this mic is that it records in a way that eliminates reverb in situations where many people are speaking from different parts of a room. PZMs are often used to record a meeting around a table, or they can be taped to the wall in the back of a room full of activity (like a performance). A lavalier can be mounted on a piece of wood or metal to create an impromptu PZM.

■ Figure 13.29 Pressure Zone Microphones (PZMs) are mounted on a plate and are useful for recording multiple voices around a table or in a room.

Onboard Microphones

Another microphone that should be mentioned is the onboard microphone or camera microphone (Figure 13.30). These micro-phones are typically of very low quality and should be avoided. Professional cameras allow you to mount your own microphone, which is fine as a backup but has disadvantages even when it is of high quality. Using an onboard mic as your only microphone can be problematic. Since often your camera will be pointing at something besides the source of the audio you want (at a cutaway, or a reaction shot, for example), it is best to allow a sound person to control microphone placement and give the cameraperson the freedom to shoot without having to consider the audio being recorded by the onboard mic. Finally, camera microphones often pick up noises from the camera operator and from camera mechanisms like the servo zoom.

■ Figure 13.30 Camera-mounted microphones are commonly found on even the cheapest video cameras, but the inability to control their position in relation to subjects makes them of limited use.

Microphone Support

While documentary filmmaking stresses mobility, there are occasions where a microphone stand can be a lifesaver (Figure 13.31). One example is that a stand can take the place of a non-existent boom operator during an interview. Mic stands come in both free-standing and desk models. Most sound kits include at least a small folding desk mic stand.

The Importance of Headphones!

It is impossible to overemphasize the importance of wearing good-quality headphones and using them to monitor your recorded audio signal. Unfortunately, this is an error we often see (or hear) in documentary production classes: a great scene with no audio, or audio that is rendered unusable by interference, hum, or some problem in the environment. Earbuds are great for listening to your tablet or phone, but for documentary production invest in a pair of high quality headphones that can isolate your recorded audio from the sound in your environment.

■ Figure 13.31 A desk stand like this Quik Lok is small, light, and can be incredibly helpful, especially for a “one-man band” shooter.