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By
Robert Orban and Greg Ogonowski
Orban/CRL
Continuous From System Considerations
In digital
signal processing devices,
the lowest number of bits
per word necessary
to
achieve professional quality is 24 bits. This is because there are a number of common DSP
operations (like infinite-impulse-response filtering) that substantially increase the
digital noise floor, and 24 bits allows enough headroom to accommodate this without
audibly losing quality. (This assumes that the designer is sophisticated enough to use
appropriate measures to control noise when particularly difficult filters are used.) If
floating-point arithmetic is used, the lowest acceptable word length for professional
quality is 32 bits (24-bit mantissa and 8-bit exponent; sometimes called
“single-precision”). In digital
distribution systems,
20-bit words (120dB dynamic range) are usually adequate to represent the signal
accurately. 20 bits can retain the full quality of a 16-bit source even after as much as
24dB attenuation by a mixer. There are almost no A/D converters that can achieve more than
20 bits of real accuracy, and many “24-bit” converters have accuracy considerably
below
the 20-bit level. “Marketing bits”
in A/D converters are outrageously abused to deceive customers, and, if these 4
Jensen
Transformer, Inc., North
Hollywood,
California,
USA
(Phone +1 213 876-0059, or Fax +1 818 7634574) 5
Lundahl
Transformers AB, Tibeliusgatan 7 SE-761 50, Norrtalje SWEDEN
(Phone: +46 - 176 139 30 Fax: +46 - 176 139 35) A/D converters were consumer products, the
Federal Trade Commission would doubtless quickly forbid such bogus claims. There is
considerable disagreement about the audible benefits (if any) of raising
the
sample rate above
44.1 kHz. To the author’s knowledge, as of 1999 there have been
no rigorous tests of this that are double-blind and that adequately control for other
variables, like the performance of the hardware as the sample rate is changed. (Assuming
perfect hardware, it can be shown that this debate comes down entirely to the audibility
of a given anti-aliasing filter design, as is discussed below.) Nevertheless, in a
marketing-driven push, the record industry is attempting to change the consumer standard
from 44.1 kHz to 96 kHz. Regardless of whether scientifically accurate testing eventually
proves that this is audibly beneficial, it has no benefit in FM stereo because the
sampling rate of FM stereo is 38 kHz, so the signal must eventually be lowpass-filtered to
17 kHz or less to prevent aliasing. It is beneficial in DAR, which typically has 20 kHz
audio bandwidth, but offers no benefit at all in AM, whose bandwidth is no greater than 10
kHz in any country and is often 4.5 kHz. There has
been
at least one rigorous test comparing 48 kHz and 96 kHz sample rates6.
This test concluded that there is no audible difference between these two sample
rates if the 48 kHz rate’s anti-aliasing filter is appropriately designed. Some A/D
converters have
built-in soft clippers that start to act when the input signal is 3 – 6 dB below full
scale. While these can be useful in mastering work, they have no place in transferring
previously mastered recordings (like commercial CD). If the soft clipper in an A/D
converter cannot be defeated, that A/D should not be used for transfer work. Dither
is
random noise that is added to the signal at approximately the level of the least
significant bit. It should be added to the analog signal before the A/D converter, and to
any digital signal before its word length is shortened. Its purpose is to linearize the
digital system by changing what is, in essence, “crossover distortion” into audibly
innocuous random noise. Without dither, any signal falling below the level of the least
significant bit will disappear altogether. Dither will randomly move this signal through
the threshold of the LSB, rendering it audible (though noisy). Whenever any DSP operation
is performed on the signal (particularly decreasing gain), the resulting signal must be
re-dithered before the word length is truncated back to the length of the input words.
Ordinarily, correct dither is added in the A/D stage of any competent commercial product
performing the conversion. However, some products allow the user to turn the dither on or
off when truncating the length of a word in the digital domain. If the user chooses to
omit adding dither, this should be because the signal in question already contained enough
dither noise to make it unnecessary to add more. In the absence of “noise
shaping,”
the spectrum of the usual “triangularprobability- function
(TPF)” dither is white (that is, each arithmetic frequency increment contains the same
energy). However, noise shaping can change this noise spectrum to concentrate most of the
dither energy into the frequency range where the ear is least sensitive. In practice, this
means reducing the energy around 4 kHz and 6
Katz,
Bob: Mastering
Audio: the art and the science. Oxford,
Focal Press, 2002, p. 223 raising
it above 9 kHz. Doing this can increase the effective resolution of a 16-bit system to
almost 19 bits in the crucial midrange area, and is standard in CD mastering. There are
many proprietary curves used by various manufacturers for noise shaping, and each has a
slightly different sound. It has been shown that passing noise shaped dither through most
classes of signal processing and/or a D/A converter with non-monotonic behavior will
destroy the advantages of the noise shaping by “filling in” the frequency areas where
the original noise-shaped signal had little energy. The result is usually poorer than if
no noise shaping had been used. For this reason, Orban has adopted a conservative approach
to noise shaping, recommending so-called “first-order highpass” noise shaping and
implementing this in Orban products that allow dither to be added to their digital output
streams. First-order highpass noise shaping provides a substantial improvement in
resolution over simple white TPF dither, but its total noise power is only 3dB higher than
white TPF dither. Therefore, if it is passed through additional signal processing and/or
an imperfect D/A converter, there will be little noise penalty by comparison to more
aggressive noise shaping schemes. One of the great benefits of the digitization
of the signal path in
broadcasting is this:
Once in digital form, the signal is far less subject to subtle degradation than it would
be if it were in analog form. Short of becoming entirely un-decodable, the worst that can
happen to the signal is deterioration of noise-shaped dither, and/or added jitter. Jitter
is a time-base error. The only jitter than cannot be removed from the signal is jitter
that was added in the original analog-to-digital conversion process. All
subsequent
jitter can be completely removed in a sort of “time-base correction” operation,
accurately recovering the original signal. The only limitation is the performance of the
“time-base correction” circuitry, which requires sophisticated design to reduce added
jitter below audibility. This “time-base correction” usually occurs in the digital
input receiver, although further stages can be used downstream. There
are several pervasive myths regarding digital audio: One
myth is that long
reconstruction filters smear the transient response of digital
audio,
and that there is therefore an advantage to using a reconstruction filter
with a short impulse response, even if this means rolling off frequencies above 10 kHz.
Several commercial high-end D-to-A converters operate on exactly this mistaken assumption.
This is one area of digital audio where intuition is particularly deceptive. The sole
purpose of a reconstruction filter is to fill in the missing pieces between the digital
samples. These days, symmetrical finite-impulse-response filters are used for this task
because they have no phase distortion. The output of such a filter is a weighted sum of
the digital samples symmetrically surrounding the point being reconstructed. The more
samples that are used, the better and more accurate the result, even if this means that
the filter is very long. It’s easiest to justify this assertion in the frequency domain.
Provided that the frequencies in the passband and the transition region of the original
anti-aliasing filter are entirely within the passband of the reconstruction filter, then
the reconstruction filter will act only as a delay line and will pass the audio without
distortion. Of course, all practical reconstruction filters have slight frequency response
ripples in their passbands, and these can affect the sound by making the amplitude
response (but not the phase response) of the “delay line” slightly imperfect. But
typically, these ripples are in the order of a few thousandths of a dB in high-quality
equipment and are very unlikely to be audible. The authors have proved this experimentally
by simulating such a system and subtracting the output of the reconstruction filter from
its input to determine what errors the reconstruction filter introduces. Of course, you
have to add a time delay to the input to compensate for the reconstruction filter’s
delay. The source signal was random noise, applied to a very sharp filter that
band-limited the white noise so that its energy was entirely within the passband of the
reconstruction filter. We used a very high-quality linear-phase FIR reconstruction filter
and ran the simulation in double-precision floating-point arithmetic. The resulting error
signal was a minimum of 125 dB below full scale on a sample-by-sample basis, which was
comparable to the stopband depth in the experimental reconstruction filter. We therefore
have the paradoxical result that, in a properly designed digital audio system, the
frequency response of the system and its sound is determined by the anti-aliasing filter
and not by the reconstruction filter. Provided that they are realized with high-precision
arithmetic, longer reconstruction filters are always better. This means that a rigorous
way to test the assumption that high sample rates soundbetter than low sample rates is to
set up a high-sample rate system. Then, without changing any other variable, introduce a
filter in the digital domain with the same frequency response as the high-quality
anti-aliasing filter that would be required for the lower sample rate. If you cannot
detect the presence of this filter in a doubleblind test, then you have just proved that
the higher sample rate has no intrinsic audible advantage, because you can always make the
reconstruction filter audibly transparent. Another myth is that digital
audio cannot resolve time differences smaller than
one sample period, and therefore damages the stereo image.
People who believe this like to imagine an analog step moving in time between two sample
points. They argue that there will be no change in the output of the A/D converter until
the step crosses one sample point and therefore the time resolution is limited to one
sample. The problem with this argument is that there is no such thing as an
infinite-risetimestep function in the digital domain. To be properly represented, such a
function has to first be applied to an anti-aliasing filter. This filter turns the step
into an exponential ramp, which typically has equal pre- and post-ringing. This ramp can
be moved far less than one sample period in time and still cause the sample points to
change value. In fact, assuming no jitter and correct dithering, the time resolution of a
digital system is the same as an analog system having the same bandwidth and noise floor.
Ultimately, the time resolution is determined by the sampling frequency and by the noise
floor of the system. As you try to get finer and finer resolution, the measurements will
become more and more uncertain due to dither noise. Finally, you will get to the point
where noise obscures the signal and your measurement cannot get any finer. However, this
point is orders of magnitude smaller in time than one sample period and
is the same as in an analog system. A
final myth is that upsampling
digital audio to a higher sample frequency will
increase audio quality or resolution.
In fact, the original recording at the original sample rate contains all of the
information obtainable from that recording. The only thing that raising the sample
frequency does is to add ultrasonic images of the original audio around the new sample
frequency. In any correctly designed sample rate converter, these are reduced (but never
entirely eliminated) by a filter following the upsampler. People who claim to hear
differences between “upsampled” audio and the original are either imagining things or
hearing coloration caused by the added image frequencies or the frequency response of the
upsampler’s filter. They are not
hearing
a more accurate reproduction of the original recording. This also applies to the sample
rate conversion that
often occurs in a digital facility. It
is quite possible to create a sample rate converter whose filters are poor enough to make
images audible. One should test any sample rate converter, hardware or software, intended
for use in professional audio by converting the highest frequency sinewave in the bandpass
of the audio being converted, which is typically about 0.45 times the sample frequency.
Observe the output of the SRC on a spectrum analyzer or with software containing an FFT
analyzer (like Adobe Audition or Cool Edit). In a professional-quality SRC, images will be
at least 90 dB below the desired signal, and, in SRC’s designed to accommodate long word
lengths (like 24 bit), images will often be –120 dB or lower. And
finally, some truisms regarding loudness and quality: Every
radio is equipped with a volume control, and every listener knows how to use it. If the
listener has access to the volume control, he or she will adjust it to his or her
preferred loudness. After said listener does this, the only thing left distinguishing the
“sound” of the radio station is its texture, which will be either clean or degraded,
depending on the source quality and the audio processing. Any Program Director who boasts
of his station’s $20,000 worth of “enhancement” equipment should be first taken to a
physician who can clean the wax from his ears, then forced to swear that he is not under
the influence of any suspicious substances, and finally placed gently but firmly in front
of a high-quality monitor system for a demonstration of the degradation that $20,000 worth
of “enhancement” causes! Always remember that less is more.
Stereo Enhancement
In
contemporary broadcast audio processing, high value is placed on the loudness and impact
of a station compared to its competition. OPTIMOD already has made a major contribution to
competiveness. Orban’s 222A Stereo Spatial Enhancer augments your station’s spatial
image to achieve a more dramatic and more listenable sound. Your stereo image will become
magnified and intensified; your listeners will also perceive greater loudness, brightness,
clarity, dynamics and depth. In use, the 222A detects and enhances the attack transients
present in all stereo program material, while not processing other portions. Because the
ear relies primarily on attack transients to determine the location of a sound source in
the stereo image, this technique increases the apparent width of the stereo soundstage.
Since only attack transients are affected, the average L–R energy is not significantly
increased, so the 222A does not exacerbate multiple distortion. Several of Orban’s
digital audio processors now incorporate the 222A algorithm in DSP.
Other
Production Equipment
The
preceding discussions of disk reproduction, tape, and electronic quality also apply to the
production studio. Compact discs and DVD-Audio discs usually provide the highest quality.
For cuts that must be taken from vinyl disk, it is preferable to use “high-end”
consumer phono cartridges, arms, and turntables in production. Make sure that one
person
has responsibility for production quality and for preventing abuse of the record playing
equipment. Having a single production director will also help achieve a consistent air
sound—an important contribution to the “big-time” sound many stations want. A new
generation of low-cost all-digital mixers, made by companies like Soundcraft, Yamaha,
Mackie, and Roland, provide the ability to automate mixes and to keep the signal in the
digital domain throughout the production process. At the high end, Orban’s Audicy
digital workstation is oriented towards fast radio production. It comin
combines
a dedicated mixing control surface with no-delay RAM-based editing and high-quality
built-in digital effects. Although some people still swear by certain “classic”
vacuum-tube power amplifiers (notably those manufactured by Marantz and McIntosh), the
best choice for a monitor amplifier is probably a medium-power (100 watts or so per
channel) solid-state amplifier with a good record of reliability in professional
applications. We do not recommend using an amplifier that employs a magnetic field power
supply or other such unusual technology, because these amplifiers literally chop cycles of
the AC power line and tend to cause RFI problems. Do not be tempted to dust off an old
Gates or RCA power amplifier and place it in service because it saves you money. It is
also usually unwise to use the monitor amplifiers built into most consoles.
Production
Practices
The
following represents our opinions on production practices. We are aware that some stations
operate under substantially different philosophies. But we feel that the recommendations
below are rational and offer a good guide to achieving consistently high quality.
1.
Do not apply general audio processing to dubs from commercial recordings in the production
studio.
OPTIMOD
provides all the processing necessary, and does so with a remarkable lack of audible side
effects. Further compression is not only undesirable but is likely to be very audible. If
the production compressor has a slow attack time (and therefore produces overshoots that
can activate gain reduction in OPTIMOD), it will probably “fight” with OPTIMOD,
ultimately yielding a substantially worse air sound than one might expect given the
individual sounds of the two units. If it proves impossible to train production personnel
to record with the correct levels, we recommend using the Orban 8200ST (an integrated
gated leveler, compressor, high-frequency limiter, and peak clipper that is the successor
to Orban’s 464A “Co-Operator”) to protect the production recorder from overload.
When used for leveling only, the 8200ST does not affect short-term peak-to-average ratio
of the audio, and so will not introduce unnatural artifacts into OPTIMOD processing. If
production personnel control levels correctly, the 8200ST can be used as a safety limiter
and high-frequency limiter by using only the 8200ST compressor function and adjusting its
input gain so that broadband gain reduction never occurs when the console VU meters are
peaking normally. With this set-up, only high frequencies will be controlled and
high-frequency tape saturation will be prevented without adding unwanted broadband
compression. (The 8200ST subtle broadband compressor will still prevent tape overload if
the console output level is peaked too high.)
2.
Avoid excessive bass and treble boost.
Sub-standard
recordings can be sweetened with equalization to achieve a tonal balance typical of the
best currently produced recordings. However, excessive treble boost (to achieve a certain
sound signature for the station) must be avoided if a tape speed of 7.5ips is used,
because the tape is subject to high-frequency saturation due to the high-frequency boost
applied by the recorder’s equalization network. If production is recorded and played
on-air digitally, there is no effective limit to the amount of HF boost you can apply in
production. However be aware that large amounts of HF boost will stress your on-air AM or
FM audio processor because it has to deal with pre-emphasis. We recommend using a modern
CD typical of your program material as a reference for spectral balance. Very experienced
engineers master major-label CDs using the best available processing and monitoring
equipment, typically costing over $100,000 per room in a well-equipped mastering studio.
The sound of majorlabel CDs represents an artful compromise between the demands of
different types of playback systems and is designed to sound good on all of them.
Mastering engineers do not make these compromises lightly. We believe it is very unwise
for a radio station to significantly depart from the spectral balance typical of
major-label CDs, because this almost certainly guarantees that there will be a class of
receivers on which the station sounds terrible.
3.
Pay particular attention to the maintenance of production studio equipment.
Even
greater care than that employed in maintaining on-air equipment is necessary in the
production studio, since quality loss here will appear on the air repeatedly. The
production director should be acutely sensitized to audible quality degradation and should
immediately inform the engineering staff of any problems detected by ear.
4.
Minimize motor noise.
To
prevent motor noise from leaking into the production microphone, tape machines with noisy
motors and computers with noisy fans and hard drives should be installed in alcoves under
soffits, and surrounded by acoustic treatment. In the real world of budget limitations
this is sometimes not possible, although sound-deadening treatment of small spaces is so
inexpensive that there is little excuse for not doing it. But even in an untreated room,
it is possible to use a directional microphone (with figure-eight configuration, for
example) with the noisy machine placed on the microphone’s “dead” axis. Choosing the
frequency response of the microphone to avoid exaggerating low frequencies will help. In
particularly difficult cases, a noise gate or expander can be used after the microphone
preamp to shut off the microphone except during actual speech.
5.
Consider processing the microphone signal.
Audio
processing can be applied to the microphone channel to give the sound more punch. Suitable
equalization may include gentle low- and high-frequency boosts to crispen sound, aid
intelligibility, and add a “big-time” quality to the announcer. But be careful not to
use too much bass boost, because it can degrade
intelligibility.
Effects like telephone and transistor radio can be achieved with equalization, too. The
punch of production material can often be enhanced by tasteful application of compression
to the microphone chain. However, avoid using an excessive amount of gain reduction and
excessively fast release time. These cause room noise and announcer breath sounds to be
exaggerated to grotesque levels (although this problem can be minimized if the compressor
has a built-in expander or noise gate function). When adjusting the microphone processor,
adjust the on-air audio processor for your desired sound on music first, and then adjust
the microphone processor to complement the on-air processing you have selected.
Close-micing, which is customary in the production studio, can exaggerate voice sibilance.
In addition, many women’s voices are sibilant enough to cause unpleasant effects.
High-frequency equalization and/or compression will further exaggerate sibilance. If you
prefer an uncompressed sound for production work but still have a sibilance problem, then
consider locating a dedicated de-esser after
all
other processing in the microphone chain.
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