(2017-05-25) Microphones
The heart and soul of audio recording.
This topic is discussed at length in a dedicated page
which introduces the fundamentals and explains the meaning of the
various characteristics which appear in microphone spec sheets.
To help you choose the right microphone, more than 150 models
are presented there to illustrate the discussion of half-a-dozen different categories.
Current prices and specifications are provided with links to specific reference pages.
(2018-01-22) "Au clair de la lune" predated "Mary had a little lamb".
From tinfoil to wax, to vinyl. From magnetic wire to tape, to digital.
Edouard-Léon Scott, 1860 : Au clair de la lune, mon ami Pierrot,
prête moi ta plume, pour écrire un mot.
Ma chandelle est morte, je n'ai plus de feu.
Ouvre-moi ta porte, pour l'amour de Dieu.
Au clair de la lune, Pierrot répondit...
Thomas Alva Edison, 1877 : Mary had a little lamb.
Its fleece was white as snow
and everywhere that Mary went
the lamb was sure to go
...
Those spoken words were the first which Thomas Edison
himself played back on his 1877 phonograph.
At the time, his tinfoil records could be played back only a couple of times
before the stylus ruined the groove.
Yet, this was a dramatic improvement on the phonautographe patented in 1857
by the Frenchman
Edouard-Léon
Scott de Martinville (1817-1879) which recorded sound graphically but couldn't play it back!
With modern technology, we can hear Scott's phonautograms now. The oldest extant
recording of a human voice is a 20-second slow rendition of
Au clair de la lune by Scott himself,
dated April 9, 1860 (unearthed in 2008, it was originally played at twice the correct speed
and was mistaken for the voice of a young woman). We also have a phonautogram of a
cornet from 1857.
The phonautograms produced by Edouard-Léon Scott were calibrated with a parallel track made by
a 125 Hz tuning fork.
This makes it possible to reproduce their sound accurately in modern times.
The team led by David Giovannoni and Patrick Feaster
originally mistook Scott's unusual indication
(500 "simple vibrations" per second) to mean 250 Hz, according to the
old practice of counting half-periods as beats.
Little did they know that
Scott was actually counting quarter-periods and meant to say he was using a 125 Hz tuning fork.
So, originally, they played back the record at twice the correct speed
and publicized their finding as a 10-second clip of the voice of a young woman...
Actually, the lone experimenter was just testing his own recording instrument
by himself in the Spring of 1860. Makes sense.
The next step was the introduction of wax cylinders to produce more permanent recordings
which could be played back many times.
(2017-11-03) Room Acoustics. Dry room, audio booth, sound stage.
Avoiding two pitfalls: Outside noise and inside echoes.
You can listen to recorded sound anywhere and you can record it anywhere.
In either case, however, unacceptable degradation will result
if a few simple precautions are not taken concerning isolation from
outside noise and prevention from echoing within the room.
One echo is not necessarily that bad in a small room; it's the reverberation
from multiple echoes that's unacceptable.
Soundproof Curtains
This cuts down on noise from the street (if you don't want the
inconvenience of plugging the windows).
Thick curtains also dampen the echoes from the room almost as well as
acoustic panels.
Consider hanging such curtains in front of a mirrored wall, if you have one
(either that, or treat the opposite wall more thoroughly).
(2018-02-11) Reverbs and Delays
Adding natural or unnatural dimension to sound, using echoes.
A reverberation unit mixes a signal with a delayed attenuated version of itself
As the result of the mix is processed the same way, we obtain multiple echoes of decreasing intensities.
There are only two control paramters which can be adjusted at will:
The delay time t and the attenuation r < 1.
As the echoes are added to the original signal, the average
volume is increased by a factor of 1/(1-r)
which is the sum of a geometric series.
(2018-02-03) Microphone Stands and Booms
By far, the most common thread for microphone mounts is 5/8''-27.
5/8''-27 means a diameter of 5/8'' (15.9 mm)
and 27 threads per inch.
In the old days, the microphones themselves were threaded.
Such was the case for the legendary Model 55 Unidyne Microphone
introduced by Shure Brothers
in 1939 and once described as the most recognized microphone in the world.
The so-called Elvis microphone was a scaled-down version of the 1939 model introduced in 1955.
This vintage look is still preserved by Shure in current "55" microphones
which encase different shock-mounted modern capsules (Super 55 and 55SH II).
With just a few such exceptions, today's full-sized microphones are rarely threaded.
Instead, they fit into threaded mic clips
or shockmounts, which provide an external isolation from vibrations.
Because microphones are so often used in close proximity to cameras,
they sometimes have to share the same threaded studs and it's useful to keep
a couple of
thread adapters
handy, to fit either 3/8''-16 or 1/4''-20.
(2018-02-07) Pop filters = Pop screens = Pop shields
Eliminating plosives and guarding from spit.
For studio mics at least, I like
cylindrical pop filters best
($10 a piece).
They're even more effective (with directional mics) if you
point the microphone at your mouth, but not your mouth at the microphone.
This way, you can look directly at a camera without hiding your face behind the
microphone (again, cylindrical pop filters are the least obtrusive ones in that situation too).
When designed by people who know their craft, all computer graphics
look as if they're lit from the upper left of the screen.
(This applies, in particular, to common action buttons.)
If at all possible, a video meant to be viewed primarily on busy computer screens
should be lit the same way (key light to the left of the camera or to the
right of the talent).
Consequently, when a studio microphone is visible at the same level as the face of
the talent, it's often best to put it to the left
of the talent so the key light won't cast a shadow from the microphone on the face.
Pop Filters (21:24)
by Mike DelGaudio (Booth Junkie, 2017-04-29).
(2018-02-04) Audio Connectors: Sizes and Pinouts.
Balanced XLR for microphones. Audio jacks: ¼'',
3.5 mm, 2.5 mm.
By definition, a socket is a plug (usually female)
affixed to a box or a panel.
A tail is a (male) jack mounted on a short piece of cable hardwired
into an appliance.
The venerable ¼'' phone jack was introduced in 1878
by George W. Coy for the first commercial
manual telephone switchboard,
installed in New Haven, CT.
For a TRS 3-line jack, use the "R" mnemonic, namely: Ring = Red = Right = Return.
(Conversely, Tip = White (or black) = Left = Send.)
Insert (Send and Return) :
In a standard (unbalanced) TRS insert jack, tip (T) is send and ring (R) is return.
Sleeve (S) is the common ground. Because inserts are used at line levels, there's little
or no need for balanced lines (which are only possible with separate send and return jacks).
Pinout of a TRS Phone Jack for Balanced Audio (1/4" or 1/8")
(2018-01-14) Volume = Perceived Loudness
Loudness normalization to broadcast standards.
Humans may perceive the loudness of identical
sound levels (dB SPL) differently according to their
frequencies.
For normalization purposes, the most commonly used calibration of that effect
(in North America, at least) is called A-weighing.
Decibels of perceived loudness, following that standard ponderation, are
indicated by the symbol dBA. This can be used to measure the loudness of various
types of sounds containing diverse mixtures of frequencies:
Ear damage is likely after prolonged exposure (8 hours).
Loud snoring
Propeller-plane flyover (1000 ft)
85 dBA
Vacuum cleaner (3 ft)
Passing diesel truck
80 dBA
Busy New-York street.
75 dBA
Vacuum cleaner (10 ft)
70 dBA
Restaurant conversation
65 dBA
Air conditioner
60 dBA
Background music
55 dBA
Soft conversation
Quiet suburb (daytime)
50 dBA
Refrigerator
Light traffic
45 dBA
Library
Bird calls
40 dBA
Urban living room
Babbling brook
30 dBA
Quiet rural area
20 dBA
Whisper
Rustling leaves
10 dBA
Calm breathing
Desert without wind
0 dBA
Conventional limit of audibility
If you double the distance to a localized sound source, you reduce its loudness by 6.02 dB
(20 log 2).
Tripling that distance reduces the loudness by almost 10 dB (9.54 dB).
Multiplying the distance to the sound source by 10 reduces perceived loudness by
exactly 20 dB.
A 1 kHz sound at 1 dB SPL is nearly 1 dBA, by definition (IEC 651).
(2018-02-12) Frequency Response & Equalization (EQ)
Bass, midrange, treble and everything in-between or beyond.
Besides volume,
the equalization controls are the most common in sound systems
(from the most rudimentary bass and high
knobs to multi-band equalizers and notch filters).
EQ Explained (5:20)
Sound Basics #2, by Stella Gotshtein (Waves Audio, 2016-11-22).
How the Pros Use EQ (36:43)
by Rick Beato (2017-01-11).
(2018-01-15) Automatic Gain Control, Limiters and Compressors
Three ways to make the best of imperfect recording setups.
The input levels of an audio recording device are best adjusted manually to
account for the different sensibilities of
microphones and widely varying recording conditions (intrinsic loudness and
distance from the sources).
(2018-01-10) Microphone Preamplifiers
External preamplifiers or built-in camera preamplifiers.
Often, you just plug your microphone into the built-in preamplifier
of a camera (to be avoided) a
portable recorder,
a mixer or some audio interface.
Stand-alone dedicated preamps are usally much better,
especially if you can bypass the aforementioned built-in preamps entirely
(e.g., using a return jack on a mixer).
If you must go through a regular audio input with built-in preamp,
the best rule of thumb to minimize noise is to set it to the lowest available setting
and adjust the external gain so that the meters peak between -18dB and -12dB.
This will keep you safely at no more than 25% of the level beyond which hard clipping occurs.
(2018-02-09) Low Pass = High Cut
A good low-pass analog filter is paramount for proper digitization.
Everything above 20 kHz is utterly useless.
The human ear is unable to detect is. Only kids can hear 20 kHz.
Young adults are lucky if they can detect a sinewave at 18 kHz.
Middle-aged people can't hear a thing above 14 kHz or 15 kHz, at best.
At the age of 62, I can still hear 10 kHz (D#9 more than one octave
beyond the range of the piano) but only sporadically beyond that.
Furthermore, with a 48 kHz
sampling rate anything above 24 kHz will actually damage
the digitized audio signal beyond repair (in the form of additional audible noise).
For the utmost in quality, we must attenuate everything in that part the spectrum
as much as possible with analog filteringprior to digital sampling.
Any leftover ultrasonic component (beyond exactly 24 kHz)
will cause audible noise in the digitized audio signal.
Do keep that in mind if you happen to use a fancy
Earthworks microphone with an unusually wide bandwidth.
Those need more low-pass filtering...
In the analog to digital conversion process,
any ultrasound translate into muddy hiss, not added clarity!
The best place to put extra low-pass filtering, if needed, is after the
microphone preamplifier and before the
ADC in the audio interface,
assuming that those two components are separate or have insert capability.
(2018-04-26) Analog-to-Digital Conversion (ADC).
Generation loss is the degradation which occurs with each pair of conversion.
(2018-04-26) Generation Loss
A degradation occurs with each ADC/DCA pair in the signal path.
Nowadays, wireless microphones use digital transmission
for greater reliability and greater fidelity. However, the signal received
is rarely kept in its digital form, if ever.
Instead, it's immediately converted back to analog form throughh a DAC.
This degraded signal is all the rest of the signal chain will ever see.
It may well be less than the degradation undergone by audio signals which are
modulated and demodulated in an analog radio transmission, but it's a degradation all the same.
A signal generation loss is usually not audible, but three or four can be.
(2018-01-17) Lavalier microphone digital wireless systems.
I'll just mention a few good alternatives (with links) and give a full review of the
Sennheiser AVX system, which I ended up purchasing (I'll give my reasons).
The actual street price is $400 but the bundled lapel mic retails for $250...
Røde also sells the NewsShooter kit
($499)
with the same receiver paired to a more flexible transmitter, featuring both an XLR
socket (with phantom power) and a 3.5 mm socket
for third-party lavalier mics.
The above price is only an estimate
of what the system would cost if it was sold without a
lavalier mic, which isn't the case
(it's actually either bundled with a $150 ME2 for $700 or with a $400 MKE2 for $900).
The system uses proprietary rechargeable batteries and, arguably, you should have
at least one extra battery for the transmitter and one for the receiver,
for an additional cost of $100 or so (there are no third-party suppliers).
Sennheiser's AVX system is the digital successor to their very popular EW100 G3 analog model.
It's the Rolls-Royce of wireless microphones;
superb user-friendly engineering at a hefty price. Extremely easy to use.
What made me buy it in spite of the cost is the small size of the receiver,
which is a perfect match for the XLR1 (whose other
XLR mic input can then be used to capture ambient sound on the other audio track).
The radio part is designed from the ground up to provide no interference from any source
in the foreseeable future.
Both units are constantly in two-way communication to maintain a clear channel
in the allotted band. All data is continuously transmitted on two separate channels
for seamless switching from one channel to the next if needed.
Up to 8 AVX systems can share the same airspace and negotiate between themselves
for trouble-free communications without any human intervention.
For good measure, the audio data is encrypted, to prevent electromagnetic eavesdropping.
Radio communications are entirely digital, using GFSK modulation
(Gaussian Frequency Shift Keying) which is to say that the digital signal
passes through a Gaussian filter before being frequency modulated
(this method allows narrower radio bandwidth; it's what Bluetooth® uses).
The audio signal is digitized with 24-bit resolution
at a 48 kHz sampling rate.
We're very far from yesteryear's one-way analog transmission of an audio signal
over a single analog FM channel selected once and for all among a dozen choices or so.
In Sennheiser's parlance, the small receiver (actually a transceiver)
is called EKP AVX and its battery is BA 20.
The body pack (SK AVX ) takes a BA 30 battery.
You need at least one extra BA 30 for prolonged use,
since the body pack cannot be recharged while in use (it's on a untethered
moving body, after all). Either battery can be recharged whether it's
mounted to the corresponding unit or not, using a standard USB-A to
USB-C cable,
from any powered type-A socket (one such cable and a small AC adapter are included).
The LED indicator (red when charging, green when fully charged)
isn't designed for color-blind people.
Also available is a handheld microphone (SKM AVX) which takes a third
kind of battery pack (BA 10) which, surprisingly,
can't be charged when mounted (unlike the other AVX batteries).
It takes 4½ hours
to fully charge a BA 10 or BA 30
(good for up to 15 hours of continuous use).
The tiny BA 20 battery can be fully charged in 1¼ h
and will power the EKP AVX receiver for 4 hours.
A 4-level battery status is provided when the left button is pressed.
A blinking alert indicates there's less than 15 minutes of battery power left.
You can power the EKP AVX with the USB cable for an unlimited time
when working tethered.
If an EKP AVX receiver is plugged into an XLR socket with
phantom power, it will turn on and off automatically (to save battery power)
by sensing the presence of power in the socket. One less switch to worry about.
The EKP AVX turns itself off about 10 seconds after it
sees phantom voltage drop. Modern cameras take some time to switch off
and there may also be a significant delay due to the slow discharge of capacitors
with little resistive load on them. All told,
an EKP AVX connected to an XLR1
(with phantom power) on a Lumix GH5
turns itself off about 13 seconds after the camera is switched off.
One benefit is that there's no loss of pairing if you reset the camera
by power-cycling it for whatever reason.
On the other hand, be aware that the GH5 turns the XLR1 off
(along with phantom power) when it's used for previewing clips
on the back of the camera. To get out of this power-saving mode,
half-press the shutter button and wait up to 10 seconds for a new
pairing to take place (the camera shouldn't be too far from the mic).
The SK AVX bodypack input socket accepts either a line input or a microphone
(including third-party replacement microphone with Sennheiser/Sony locking jacks).
The bandwidth for a line source is 20Hz to 20kHz. For a microphone, it's
limited to 50Hz to 20kHz (which is more than enough).
The
Sennheiser
EW plug (Evolution Wireless) is a 3.5 mm locking audio plug with
TRS
connections (tip-ring-sleeve).
It's used for audio input from either a microphones (tip) or line signals
(ring; 1 MW input impedance).
Whichever input is unused must be grounded (i.e., connected to the shielding sleeve)
within the input plug and/or the input device.
The AVX system has a constant latency of 19 ms
which would correspond to a sound source located 6.5 m (21 ft)
away from the listener. There's usually no need to adjust that in post-production,
except possibly for an extreme close-up shot of a person talking, in which case
reducing the latency ought to be reduced down to 1 ms or 2 ms
(never less) to reproduce more precisely the time-delay our brains are accustomed to
when carrying a conversation up close (1 ft or 2 ft away).
most accustomed to.
Likewise, if you have to synchronize sound based on the image of a
clapper, make sure you introduce a delay corresponding
to a delay of about one millisecond per foot of distance between the camera and the subject
(the brain will effortlessly compensate for slightly more delay but will be confused by less).
Future Proof? Worldwide usage?
Sennheiser chose to use the 1900 MHz which is currently relatively free of interference
from competitive devices. This is much less crowded than the 2400 MHz band.
(2018-01-10) Panasonic's XLR1 adapter for the Lumix GH5
The pricey ($400) XLR1 is a key audio accessory for the GH5 camera.
The XLR1 bypasses entirely the regular audio input of the GH5
and communicates digitally with the camera via the hot-shoe contacts.
The only other way to achieve the same audio quality is to use a good
external recorder to produce an independent soundtrack for later synchronization...
Turn the camera off before connecting or disconnecting the XLR1
to the hot-shoe. (I don't think you could damage the camera by ignoring this
recommendation but this would definitely confuse the software.)
Mounting the unit disables the single 16-bit microphone input of the camera
(unless the user chooses to disable the XLR1 by software).
The XLR1 contains a pair of audio preamplifiers and 24-bit converters.
To save battery power, the XLR1 will turn itself off when the
camera goes into viewing mode. That will turn off the connected
devices which depend on 48 V phantom power or those which merely
sense it, including the Sennheiser AVX
wireless microphone system (after exiting viewing mode, allow 10 seconds
for the AVX system to properly re-establish its radio link).
If you absolutely can't leave with that, give up the AVX auto-off feature
by not feeding it phantom power at all (this will force you to turn
the AVX receiver on and off manually).
(2018-05-03) Evaluating Audio Compression Curves
The THD of a recorded sine wave is a function of the compression ratio.
To prevent hard clipping, a signal u(t) is recorded as f (u(t)) as
time (t) varies.
In this, f is always chosen so that its values remain between the output rails
(normalized to -1 and +1).
The quality of the compression for a given amplitude x must be defined to be a decreasing function the
total harmonic distortion (THD) of the signal f (x sin u(t)).
We may choose it that to be the linear function which is 0 fot the THD of a square wave
(the hardest clipping case) and 1 (100%) for a THD of 0 (undistorted sine wave).
The problem is to abandon the requirement of 100% quality for small signals in order to
handle with reasonable distortion larger signals. This trade-off will utimately depend on
a single user-defined parameter which determines the position of the "knee".
For a given position, the shape of the knee can be optimized.
(2018-02-16) DI Boxes (Direct injection or Direct insertion)
Active and passive DI boxes fetch clean instrument signals directly.
Generally speaking, you should use an active DI box (requiring phantom power)
with a passive instrument and a passive DI box with an active instrument (unless the output is really weak).
Some modern active DI boxes are designed with widely separated voltage rails
(9V in the case of Radial's RJ48) which allows them to be used also with strong signals without clipping.
Typically, DI boxes are placed on the way from an electric guitar to its amplifier and provide a secondary
low-impedance balanced output line.
That can be directed to an audio interface to record the
output of the guitar cleanly, without the tonality added by a specific amp (which is what a
microphone would record). The process by which a DAW gives such a clean
track the coloration of some virtual amp (chosen in post-production) is called re-amping.
(2018-02-05) Audio transformers with male plugs:
Step-up transformers are used to match microphones to high-Z inputs.
A passivedirect box is essentially just
a step-down audio transformer. Conversely, a step-up transformer ("lo to hi") can be used
to match a microphone output to a high-level high-impedance input, like that of a guitar amp.
Being just a transformer, a passive direct box can be used backwards as a step-up transformer
to match a microphone to a hi-Z input. However, a DI-box only has a male XLR output which can
be used as input only with a gender change (for example, using a female-female XLR cable).
Adapters are available from various manufacturers, with
a low-impedance input on an XLR female socket and a high-impedance output on a mono
phone jack
(either 3.5mm for a video camera or a laptop, or ¼'' for
PA or guitar amps).
For field use,
it's mechanically good to have the phone jack mounted on a short cable (i.e, a tail).
(either that or build a small extender yourself with a female jack and a male jack, which could be either straight or cornered).
In a transformer the (inductive) impedancce Z of either side is proportional
to the square of its number of windings.
(HINT: The voltage is proportional to that number and the
current is inversely proportional to it.) The gain
(the ratio of the voltages) is thus the square root of the impedance ratio.
Expressed in decibels, this gives:
G [dB] = 10 log ( Zout/ Zin )
Note that a passive impedance transformer can't be used to raise the
voltage level into an amplifier of low input impedance, because that
low impedance would produce a collapse of the signal from
a high-impedance source which would cancel the gain produced by a passive
impedance transformer.
This misconception is so common that Shure saw fit to put a warning to
that effet on the package of the aforementioned A85F unit.
The two audio tracks (left and right) of a video are simply not enough
to solve all recording situations.
In some cases, it's not even an option
(no audio is recorded with variable frame-rate (VFR)
slow-motion footage.
An external digital audio
recorder adds considerable flexibility to common recording situations.
Handy Recorders from Zoom
All of Zoom's "H" handy recorders share the same high audio quality going
from compressed MP3 to CD-quality
(16-bit resolution,
44.1 kHz sampling rate)
and video-track standards from 16-bit 48kHz to 24-bit 96kHz.
Chronologically, Zoom introduced the H2 first, in
2007.
The popular entry-level H1 was released in
2011.
It fits the needs of most video bloggers.
The news from the CES show in Las Vegas (January 2018) is that the H1 is being discontinued and
replaced by a new model, the H1n, which will be widely available this month.
The new H1n retails for $120 while unused H1 units are still available on Amazon for $70,
while they last:
The H1's custom amber LCD is replaced by a bluish (96 by 64) dot-matrix
LCD of the same size (1¼").
The front panel of the H1n has controls
formerly located on the side or rear panels of the H1.
There's now a dedicated dial to control input level.
The H1's rear sliders have been replaced by buttons
whose status is updated on the bottom line of the LCD.
The H1n counter is able to display hundred of hours;
32GB in MP3 is up to 555 h 33 min.
(Early H1 firmware was limited to 2GB files; at most 34 h 43 min in MP3.)
New features include a switchable limiter and several low-cut filters.
The H1n now uses two AAA batteries
(the H1 used one AA). Rechargeable batteries recommended.
They both feature the same proprietary 10-pin extension connector as the H5 and H6 which allows either
two additional microphone XLR inputs (without phantom power) or one of their own microphones
(preferably with an ECM-3
extension cable).
In February 2018, Zoom introduced nominally in their F-series,
a model which might be better classified with their H-series of handy recorders
(but they didn't dare call it H0).
The Zoom F1 is a very
compact digital stereo recorder without any built-in microphone.
Instead, it has the same 10-pin proprietary port as the H5 and H6 and can
accept the same capsules.
Either that or it can use a 3.5mm microphone/line input, like the H1 or H1n
(especially for lavalier mics).
The F1 is available in two different bundles:
(2018-01-07) On the original Zoom H1 ultra-portable recorder:
Tips for setting up and using the Zoom H1 handy recorder.
With the Zoom H1, I highly recommend the APH1 accessory pack
($20) which includes
a nice branded protective case, mini-tripod stand, AC adapter, USB cable,
foam windscreen & conical adapter for standard mic clips.
(The APH1n, for the Zoom H1n, has a different protective case.)
Right-angle adapters
($6.25 a pair,
$6.99 for three)
are a must for the microphone input and/or headphone output, if you want to put the unit in your pocket
(secured with Gaffer tape or an elastic band around the whole thing).
Right-angle cables
($6.99 a pair)
are an even better and slimmer option. The cables from Cable Creation
are the best as their right-angle slim connectors will be flush with the unit, which is highly desirable.
For the Zoom H1 to generate accurate time stamps, you must first
set the time and date:
This is accomplished by holding the red button before you turn the unit on.
That way, you can access the six successive components of the date and time by pressing the
play button and change the blinking number up or down by pressing either
forward or rewind.
Next thing you have to do is choose your recording format.
Select WAV on the bottom of the unit for uncompressed recording
(compressed MP3 is only useful for cramming many hours of stand-alone sound on the micro SDHC card).
Then pressing either forward or rewind
allows you to select one of 6 choices; the combinations of 3 sampling rates
(44.1kHz, 48kHz and 96kHz) and 2 resolutions (16-bit or 24-bit).
If you intend to create video soundtracks, forget 44.1kHz (this is exclusively for audio-only CDs).
96kHz is a definite overkill, both in theory
(Nyquist-Shannon theorem)
and in practice
(you end up discarding the extra information anyway on current delivery platforms).
Therefore, consider only 48/16 or 48/24. I use exclusively the latter mode
myself, for the utmost in quality (lower digital noise, greater dynamic range
and perfect match with the best digital video formats).
This is consistent with the quality of Panasonic's XLR1
audio interface and with Sennheiser's AVX wireless system.
Even if the ultimate goal is to deliver CD-quality sound (16-bit) it's good
to record at a higher resolution. In theory, the extra 8 bits correspond
to an additional 48 dB in dynamic range, for a grand total
of 144.5 dB. The range of some microphones is up to that.
In post-production and later compression, at most a 96 dB
portion will be used, but you rarely know which part it will be...
So, keep some headroom and always record at 24-bit & 48 kHz.
Just make absolutely sure that you never clip while remaining well above the noise floor.
No hiss, no clipping and let the chips fall where they may.
With a 32GB card, the Zoom H1 can record 30 hours 46 minutes and 11 seconds of
audio at 48 kHz in 24-bit resolution. (The 2GB card bundled with the unit is only good for
115 minutes and 23 seconds.)
Once this setup is done, once and for all, the use of the unit is extremely simple and intuitive,
with or without the help of its
Quickstart Guide
(which is actually identical to the official
operating manual).
The only delicate part, as usual, is to properly adjust input levels manually.
There's a substantial degradation in sound quality if you trust AGC,
which is best reserved to special situations (like low-quality recording questions from an entire classroom
with a single fixed microphone).
Besides creating high-quality audio clips in WAV format,
I also use the H1 to record voice memos in highly-compressed MP3
(which still sounds much better than the dictating machines of yesteryear).
Since the unit remembers what type of WAV and what type of MP3 is preferred,
I can switch between the two with the flick of one finger.
When using the files, there's no question about the nature of the contents
because the file types are different.
Note that pressing play during recording creates a mark.
You can jump back and forth to such marks during playback by using rewind
and forward. Each audio file can have up to 99 such marks.
For best results, set the input level manually to 37 or more
(the equivalent input noise is higher at 36 or below). Increase the level until
the audio peaks at -12dB or so on the H1 meters. If you need to go well above 60 to
do so, your input signal is probably too weak. If you need to go below 37,
it's probably too strong. In either case, consider changing the volume of
the input device itself, if you have any way to do so (or else, there's no great
harm in going outside of the optimal 37-59 range).
Out of the box, the H1 input level is set at 50.
With firmware 2.0 and above (2013) the Zoom H1 can also be used as a card reader or a USB
audio interface. Zoom's corresponding supplement to the H1 user manual is
mirrored here.
Just connect the unit to a computer via USB before turning it on.
The display will alternate between "Card" and "Audio"; press the red button when
your desired choice is displayed (if no selection in made within 10 seconds,
the unit defaults to a card reader).
In practice, once you put a 32 GB card into this unit (that's the largest SDHC capacity)
and format it, you'll never have to remove it.
Just connect the computer with a USB cable to fetch files and erase them (or you may format
the card each time you start a new job, by pressing trashcan while turning the unit on).
The H1 derives its power from the USB connector if available and you can run it indefinitely this way,
without ever draining the battery (don't expect it to be recharged, though).
In January 2018, when the H1 unit was being discontinued, the latest version of the firmware was 2.10
(displayed as 2/10 upon any ordinary power-on).
If you need that final (?) firmware update, the relevant update file is
mirrored here. Put that file
(H1MAIN.bin, 852,224 bytes) at the root of a micro SD card and power up
the H1 with that card in it; you'll be asked to confirm that you really want to perform the update.
(See user's manual.)
(2018-02-08) The Zoom H5 four-channel handy recorder:
For $270,
the Zoom H5
delivers more recording capabilities than most small indy producers need
(just add a couple of cheap single-channel units for those rare occasions where
sound is needed from widely separated sources, or when the H5 itself is deemed too bulky).
The audio quality is great, although some quieter preamplifiers are available.
Phantom power (12V, 24V or 48V) can be brought to either of the two built-in inputs
(female combos accepting either XLR or quarter-inch jacks). The inputs on the optional
EXH-6 head
($70)
which may replaces the standard XY stereo microphone cannot
provide phantom power (they're thus suitable only for dynamic microphones,
self-powered microphones or line-level connections).
(2018-02-12) Audio interfaces (between audio signals and computers).
Sound cards, USB interfaces and USB microphones.
Two components of an audio interface deserve special consideration: The
microphone preamplifier(s) and the analog-to-digital converter(s).
The number of (digital) channels of an audio interface is the number of its ADCs.
An audio interface is mostly confined to studio work, since it's directly connected to a computer, which
is usually not as mobile as other options:
In the field, you normally use either a dedicated sound recorder
or the specialized audio interface of your video camera (either built-in or external).
Sound Cards:
A sound card fits inside a desktop computer and provides it with analog audio
inputs. Just like the built-in microphone socket of a laptop,
this has the disavantage of exposing sensitive microphone signals
to the noisy electrical environment of a computer.
Current audio gear avoid that by processing all analog signals
in an outside shielded enclosure which communicate only digitally with the main computer.
USB Microphones:
For greater simplicity and lower cost, beginners often settle for
an all-in-one solution which provides three hardwired components in a single enclosure:
An analog-to-digital converter (ADC) and a USB digital interface.
Commonly called USB Michophones,
such contraptions offer the utmost in simplicity (there's just one USB cable to hook up)
but they only provide a fixed single-microphone configuration which can't be upgraded.
USB Interfaces:
By contrast, a USB interface can convert to digital signals
the analog audio signals form several sources. They usually include at least
one microphone preamplifier, but the better ones also have insert
inputs which bypass all internal preamps and allow the use of external
preamplifiers and/or analog signal processors
(the dbx286schannel strip is popular).
In Behringer models, only Midas preamps are recommended.
The Audient-iD14 costs twice as much as the comparable top-selling Focusrite Scarlett 2i2 but it has better
mic preamps and also allows an ADAT extension of up to 8 additional digital channels.
A USB interface yields as many digital channels (over a single USB cable) as it has analog inputs.
Some rare interfaces (known as USB mixers) also integrate an analog mixer which
irrevocably combines several inputs into one channel.
One advantage of this combination is that it allows fast real-time reverbs ans echoes at little more
than the cost of an extra digital-to-analog converter (DAC).
Sound slates used to be called clappers or clapperboards.
Originally, they consisted of hinged sticks nailed on black chalkboards.
Now they're white or translucent
acrylic
plates on which dry-erase markers are used.
Magnets help the sticks snap shut.
A full-size slate is about 9½" by 12"
(bargain size: 8" by 10").
A smaller version, used for tight shots,
is called an insert slate and measures about
6" by 6", including marker sticks.
There's still no better way to match video and audio
clips recorded separately. Even when
timecodes are used,
an old-school slate helps identify everything.
Below is a summary of how professionals go about it.
Slating is normally the responsibility of the 2nd AC
(second assistant-cameraman) also called clapper-loader,
for that reason. (The 1st AC is the
focus puller.)
The second AC also maintain a camera record.
Sound recording is always started first (film used to be expensive).
The word speed is yelled to confirm that sound is being recorded
(in the old days, it used to take a short while for the reels of the recorder to reach
their operational speeds).
If there are several shots in a scene (different angles, different lenses, etc.)
they are identified
by one of 23 letters (skipping I, O and S, which could be mistaken for numerals).
Use two such letters if there are more than 23 shots, starting with AA, AB, etc.
It's best to use the
radiotelephony spelling alphabet
(Alfa, Bravo, Charlie...) or any
older variant
thereof (Able, Baker...).
Funny alternatives are sometimes improper.
The information on the slate should be accurate and updated for each "take".
The clapper is always brought into the frame in the open position and the essentials
are read (shot ID and take number, at least) the actual clap is preceded
by the word marker or mark. (Possibly preceded
by the identifier(s) of the camera(s) involved.
E.g., "A mark", "A and B mark", "A and B common marker", etc.)
Some graceful recoveries from slate snafus include:
AFS: After false start. This acronym is placed under the take number
(and pronounced aloud) when a shot is interrupted for any reason and restarted from scratch.
Second stick : The term identifies a second marker performed when a
the camera or an audio recorder missed the first one.
Tail marker : Those words precede a marker done with an upside-down slate
at the end of a shot if there was no marker at the beginning.
Put the slate right-side up after the clap.
When the slate is used just to identify the image (without associated sound)
it's held between the sticks,
In North America (not the UK) such a thing is called
MOS,
an acronym for one of many equivalent meanings:
The ultimate expression of slate etiquette is the soft stick call,
which is used to indicate that the clapper won't be used with full force out of respect
(for example, the slate may be very close to an actor's face in a quiet scene).
A dog-clicker produces a loud click which trainers use as an audio feedback for dogs and other pets.
It's also a great instrument to speed-up voice-over editing with a DAW.
The trick is to click after each fumble to clearly mark where the fumble ends and the do-over begins.
The sharp waveform of the clicker is easily recognized visually and it has to be removed
from the final cut along with a short segment that precedes it.
The beginning of that segment matches what comes just after the click mark.