0dBFS+ Levels in Digital Mastering

SØREN H. NIELSEN AND THOMAS LUND

TC Electronic A/S
Sindalsvej 34, DK-8240 Risskov, DENMARK
soerenn@tcelectronic.com, thomasl@tcelectronic.com

A sine tone at 0dBFS is often believed to be the maximum level obtainable from
a digital medium. Therefore it is typically the maximum level digital filters and
analog circuitry in consumer equipment is aimed at reproducing.
As we have showed in previous papers, inter-sample peaks may be considerably
higher than 0dBFS.
This paper examines the sonic consequences when 0dBFS+ signals are
reproduced in typical consumer equipment. The performance of a variety of
domestic CD players exposed to such signals are presented and evaluated.

0. INTRODUCTION
Several golden ears in the pro audio industry tend to believe that the best sound in pop / rock music
generally was produced between 1982 and 1995.
Despite higher resolution in converters and DSP, lower jitter and probably a better overall understanding
of digital media, we seem to be on a declining rather than inclining sound quality slope these years; even
though people buying records and film may not be aware of it.
Obviously there could be many reasons for this we cannot directly influence: Trends, basic recording and
microphone placement skills, more semi-pro equipment being used, shorter production times and
therefore less attention to detail etc.
But if the public do not care, why should we?
Because pride in our industry, craftsmanship and conservation of talent tell us to be concerned. And
because more bits, more resolution and more channels can only be justified by the end quality and listener
involvement going up.
Being a supplier of equipment for professional music and film production, TC Electronic therefore has a
continued interest in discussing goals and rules for the production and mastering process with a
pronounced focus on quality.
In this paper we have investigated millennium sound quality from a level point of view. Even though
distortion in a linear digital audio system is generally lower at high levels, there may be situations where
domestic equipment is not capable of reproducing hot signals created in a mix or mastering process.

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As we have shown in our paper at the 107th convention, Level Control in Digital Mastering [1], such level
peaks are readily derived from mastering tapes conforming to normal rules of permitting a number of
consecutive samples at full scale, or even without a single sample hitting 0 dBFS.
Areas in which 0 dBFS+ levels could be of concern are discussed, including professional equipment for
domain and sample rate conversion, data compression encoders and decoders plus, most noticeably, end
user reproduction equipment.
We will disclose our findings of how various consumer CD players react to hot levels and discuss if this
should have an influence on how digital audio is measured and mastered.
We will also investigate whether signals fulfilling the sample rate criteria versus artificial or DSP
generated signals should be looked upon differently, or if common measurement guidelines and principles
could be attained.
Finally we will discuss if the findings give us reason to continue to work on a DRA (Dynamic Range
Approval) draft as suggested in the previous paper.

0.1 Definition of 0dBFS+ Level

In the digital domain the peak level may deviate from the peak level in the analog domain.
There are two main reasons for this [1]:
1. Basic sampling theory. Sampling occurs at regular intervals, and at frequencies near integer fractions of
fs, such as fs/4 and fs/2, the phase of the signal compared to the sampling times may generate a digital
peak value somewhat below the analog peak value. If the signal is not exactly at one of the critical
frequencies mentioned above, the peak value in the digital domain will get very close to the analog peak
value. If the analog signal prior to sampling was properly bandwidth-limited, the output after digital to
analog conversion will be substantially equal to the analog input signal.

Figure 1 Figure 2
11025 Hz sine waves at full scale (0 dBFS) sampled @ 44.1 kHz.
Example shows consumer CD player, NAD C 520, measured on LeCroy 9350A digital oscilloscope.
Figure 1: Starting phase of 90°. Analog and digital peak values are identical.
Figure 2: Starting phase of 45°. Analog peak value should be +3 dBFS. Notice the clipping.
2. Gibb's phenomenon. Occurs when limiting the bandwidth of a wide-band signal (or truncating an
impulse response). This is particularly important when the signal is clipped in the digital domain, but it

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applies generally. What happens is that a square wave (or hard clipped signal) can be viewed upon as a
sum of individual sine waves of frequencies 1, 3, 5,... times the fundamental frequency. The flat top of the
square wave depends on the presence of all harmonics at the right levels and phases. If some of the
harmonics are removed by lowpass filtering, the peak value of the signal rises. When converting from
digital to analog a low pass filter is always applied, so the analog level may be higher than expected.

Figure 3 Figure 4
5512.5 Hz square waves sampled @ 44.1 kHz.
Example shows consumer CD player, NAD C 520, measured on LeCroy 9350A digital oscilloscope.
Figure 3: -6 dBFS. Notice the dip at the flat top due to lowpass filtering at the output. Only the third
harmonic fits within 22.05 kHz.
Figure 4: Digital full scale. Notice the clipped peaks. They are supposed to be twice as high as in Figure
C but clearly there is no level capabilities above a full scale sine wave.
In this paper we will refer to signals that may typically be reconstructed at a higher peak level than a sine
wave asynchronous to the sample rate for “0dBFS+” levels.

2. CRITICAL ELEMENTS OF THE SIGNAL PATH

We have not systematically investigated how performance is affected in all the types of processing
depending on intersample level computation, but many elements of the audio signal chain may be affected
by 0dBFS+ levels. E.g. filters, asynchronous and synchronous sample rate converters, data-compression
and data-expansion circuitry etc.

2.1 D/A Conversion

Our scope was to investigate consumer reproduction equipment so most of our attention has been given to
the digital to analog conversion process.
In a digital to analog reproduction chain there are several filters - and other limiting factors.
The most basic D/A converter is rarely used for audio in modern equipment, but may actually have an
advantage when it comes to behavior at high levels. The signal is sampled out of the D/A chip at the basic
sample rate, so no filtering takes place in the digital domain. An active analog output reconstruction filter

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is often run at +/-15 V supply which in most cases gives plenty of analog headroom. The fact that analog
reconstruction filters typically have non-linear phase may or may not be a disadvantage.
Many early D/A stages use a low oversampling factor like 2, 4 or 8 combined with a digital reconstruction
filter before the samples are output at this higher rate through a conventional D/A chip. In the analog
domain a simple filter attenuates mirror signals around the oversampled Nyquist frequency.
A modern D/A converter stage typically consists of one chip using a very high oversampling rate and a
built-in digital filter for reconstruction. Also with this type of converter a simple analog filter is removing
unwanted images at the output.
To summarize, these are the sources of distortion and clipping in the D/A conversion process and the
subsequent analog signal path:
1. Digital filter before the D/A converter
2. The D/A converter chip, especially the output stage
3. Analog gain stage after the converter, including AC coupling
4. Gain adjustment circuit
5. Analog output stage, possibly limited by supply voltage or current driving capability
When low pass filtering takes place in the digital domain special care must be taken to avoid overload due
to critical input signals. Also the AC coupling in the analog domain may generate high peaks - up to 6 dB
when fed by a square wave.

3. TEST AND PROCEDURE

Several different domestic and professional CD players have been submitted to a variety of test signals
constructed to reveal difficulties regarding level handling, analog circuitry lock-up etc. Test signals
include tones at different levels, program like material and artificial signals.
To check the headroom capacity of different consumer CD players, CDs were made with a collection of
test signals that could reveal problems in digital filter or analog signal-path design.
Measuring equipment consisted of LeCroy 9350A, AP System 2 Cascade and Prism DScope.

3.1 Test Signals

Signals entering a digital signal processing system from the analog world should be limited in bandwidth
and be free of aliasing. Metering of analog inputs stages is often based on the digital samples [3], [4], [5]
although some DAT recorders use analog based meters. When digitally based meters are used the analog
input gain can be set to avoid overload in the digital domain. Some obscure bandlimited signals may have
a higher amplitude in the analog than in the digital domain. Most signals, however, will be well-behaved.
In the digital domain some processing of the signal will be done, and some of it may be non-linear. In
mastering much of the processing can be non-linear: Expanding, compressing, limiting and clipping. But
also linear processing like filtering takes place: Equalisation and DC removal.
It is therefore relevant to investigate the response to simple bandwidth-limited signals as well as complex
ones which do not stick to Nyquist.
Three categories of test signals were used:
1. Sine waves
2. Square waves
3. A pseudorandom sequence

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Sine waves of four frequencies were used: 997, 5512.5, 7350 and 11025 Hz. The first frequency has no
simple relationship to the samplerate of 44100 Hz whereas the three others are fs/8, fs/6 and fs/4
respectively.
Sine waves with a simple relationship to the sample rate can be sampled in a way that the analog peak
value is substantially larger than the digital peak value.
The square wave signals can be divided into two categories with specific purposes. Two low frequency
square waves, 20 and 50 Hz, can be used to test the behavior of the AC coupling at the output. Although
only rarely full scale square waves will be generated by a signal processing algorithm something similar
will be the result if bass guitar or bass drum is clipped in order to maximize loudness. Even these low
frequency signals with only low level harmonics above the Nyquist frequency will show some output
filter ringing at the edges.
The remaining square wave signals of 551.25, 5512.5, 7350 and 11025 Hz are all of frequencies with
simple relationships to the sample rate. This is primarily done in order to avoid problems like asymmetry
and jitter of the digitally generated signals. These square waves have significant harmonic components
above the Nyquist frequency, so some overshoot due to Gibb's phenomenon must be expected in the
output filtering process. Like the low frequency square waves they are unlikely to occur in clean form in
real material but clipping does generate signals with flat tops.
As an extreme signal a pseudorandom sequence has been chosen. It is a sequence repeating every 32767
samples, consisting of only +1 and -1 (or appropriately scaled values). The frequency spectrum is white.
The peak level in the analog domain is about 6 dB higher than in the digital domain so this signal will
push filters and converters to their limits.

Figure 5 Figure 6
Pseudorandom signals sampled @ 44.1 kHz.
Example shows consumer CD player, NAD C 520, measured on LeCroy 9350A digital oscilloscope.
Figure 5: -6 dBFS amplitude in the digital domain. Notice that peaks may be around 6 dB higher than the
raw signal amplitude.
Figure 6: 0 dBFS amplitude in the digital domain. Notice that no peaks reach +6 dBFS as they
theoretically should. They are clipped.
All signals were 30 seconds in length with a half-cosine envelope at both start and end - suitable for
making simple time domain inspection as well as spectral analysis.

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3.1 Phase and Level
The synchronously sampled sine waves were generated with two start phases: One with the theoretical
maximum value present as a sample value in the digital domain and one with the highest possible analog
peak level within the limitation of +/-1 in the digital domain.
5512.5 Hz: 90 and 67.5°. At 67.5° the analog peak level is up to +0.69 dBFS.
7350 Hz: 90 and 60°. At 60° the analog peak level is up to +1.25 dBFS.
11025 Hz: 90 and 45°. At 45° the analog peak level is up to +3.0 dBFS.
The CDs made for testing contained the signals at several levels including these which have the maximum
analog peak level. As the present investigations are concerned about changed system behavior at high
levels and not about general system performance a relative measurement may be sufficient. If advanced
test equipment is available a spectral analysis will show detailed information. But a simple oscilloscope
can also tell a lot.
The main test signal is recorded on one of the two stereo channels on a CD. The other channel contains
the same signal but attenuated by 6 or 12 dB so that overload should not occur in that channel. By using a
mixer or an oscilloscope with different gain in the two channels combined with a subtraction (phase
reverse) feature the error signal can be heard or seen directly.

4. RESULTS
Sample rate synchronous sine waves were not chosen because of their program-like constitution but
because they reveal level handling limitations using conventional distortion analyzers and enable easy
comparison between units.
As a reference for the distortion measurements, asynchronous signals were used as shown below.

Figure 7 Figure 8
997 Hz @ 0dBFS sampled @ 44.1 kHz. 0 dB = Reference level for Sine Distortion Tests.
20 Hz to 100 kHz FFT on AP Cascade.
Figure 7: Yamaha CDX390 consumer CD player
Figure 8: Sony D50 consumer CD player (portable)

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4.1 Sine Test Results

A fuller picture can be obtained looking at FFT’s as shown in Fig. 9-20, but a condensed comparison is
shown in the table below.
Bandwidth of the THD+n measurements is 20 Hz - 80 kHz.

997 Hz sine 5512.5 Hz sine 7350 Hz sine 11025 Hz sine

Peak = 0.0 dBFS Peak = +0.69 dBFS Peak = +1.25 dBFS Peak = +3.0 dBFS

Denon DCD725 -61.3 dB -34.8 dB -27.0 dB -18.1 dB

Marantz CD4000 -58.8 dB -36.6 dB -30.7 dB -20.7 dB
NAD 514 -74.3 dB -30.6 dB -24.9 dB -17.2 dB
NAD 520 -67.9 dB -30.4 dB -25.8 dB -19.3 dB
Sony C11 -78.1 dB -30.2 dB -24.6 dB -16.8 dB
Sony D50 -82.9 dB -65.0 dB -59.3 dB -29.0 dB
Yamaha CDX390 -70.9 dB -33.9 dB -26.4 dB -18.3 dB

Table 1. THD+n comparison

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Figure 9 Figure 10

Figure 11 Figure 12

Sony D50 consumer CD player (portable). Analog output performance.

Sampled @ 44.1 kHz. 0 dB = Reference level for asynchronous sine wave.
Figure 9: 5512.5 Hz Sine. Theoretical analog level at 0 dBFS
Figure 10: 5512.5 Hz Sine. Theoretical analog level at +0.69 dBFS
Figure 11: 7350 Hz Sine. Theoretical analog level at +1.25 dBFS
Figure 12: 11025 Hz Sine. Theoretical analog level at +3.0 dBFS

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Figure 13 Figure 14

Figure 15 Figure 16

Denon DCD 725 consumer CD player. Analog output performance.

Sampled @ 44.1 kHz. 0 dB = Reference level for asynchronous sine wave.
Figure 13: 5512.5 Hz Sine. Theoretical analog level at 0 dBFS
Figure 14: 5512.5 Hz Sine. Theoretical analog level at +0.69 dBFS
Figure 15: 7350 Hz Sine. Theoretical analog level at +1.25 dBFS
Figure 16: 11025 Hz Sine. Theoretical analog level at +3.0 dBFS

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Figure 17 Figure 18

Figure 19 Figure 20

Sony C11 professional CD player. Analog output performance.

Sampled @ 44.1 kHz. 0 dB = Reference level for asynchronous sine wave.
Figure 17: 5512.5 Hz Sine. Theoretical analog level at 0 dBFS
Figure 18: 5512.5 Hz Sine. Theoretical analog level at +0.69 dBFS
Figure 19: 7350 Hz Sine. Theoretical analog level at +1.25 dBFS
Figure 20: 11025 Hz Sine. Theoretical analog level at +3.0 dBFS

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5. LEVEL CONTROL IN MASTERING
Level measurement in CD production has historically been a matter of counting consecutive samples at
digital full scale, 0 dBFS. Master tapes may be rejected if they contain too many consecutive samples at
full scale.
Rules based only on the number of consecutive samples clipped are useless if we want to prevent 0dBFS+
levels from occurring because it is easy to subtract a few LSBs from the signal every time full scale is hit.
LSB-cheating clearly does nothing to reduce neither level nor distortion.

5.1 Monitoring
Monitoring in most mastering studios is done using expensive stand-alone converters or mastering devices
where distortion associated with 0dBFS+ levels may be less pronounced or not exist at all.
Under such circumstances the engineer will not stand a chance to find out if the consequences could be
listening fatigue or even unmasked distortion at the end user.

5.2 Mastering against -3dBFS

Discussing deteriorating sound quality with mastering engineers, we discovered that some of them had
started using a conservative level approach several years ago. Because of experiences from the real world,
their ears had told them to generally keep peak levels below -3dBFS. Bad experiences with on-air signals
or domestic reproduction equipment may be the reason why.
Before we get more precise level monitoring tools or new level guidelines for what is allowed before a
master tape is rejected, this conservative approach certainly seems appropriate in order to avoid the kind
of consumer equipment distortion we have described in this paper.

6. CONCLUSION
All of the domestic CD players investigated have shown difficulty dealing with 0dBFS+ levels that can
easily occur on modern CDs. New models are actually worse than older types relying less on
oversampling and more on analog filters.
We have not investigated how seriously audio quality is subjectively affected, nor have we made any
listening fatigue tests concerning 0dBFS+ levels. However, modern CDs contain these kind of signals and
modern CD players are not designed to reproduce them without distortion.
There appears to be plenty of reasons for concern about the quality of audio when hot mastering levels are
to be reproduced at the end listener.
To make things worse, the mastering engineer is neither able to hear nor see when the level danger-zone is
reached.
Regardless of whether Dynamic Range Approval guidelines are adopted by the industry or not, visual
inspection tools aimed at 0dBFS+ detection should find their way into the quality-conscious mastering
studios.
It is our belief that findings like this stress the need for a continued development of a DRA. In general it
would appear that more focus should be given to how the upper end of the digital recording and
reproduction dynamic range is utilized.

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REFERENCES
[1] Søren H. Nielsen & Thomas Lund (1999): Level Control in Digital Mastering, Presented at the 107th
AES Convention, Preprint 5019.
[2] E. Zwicker & H. Fastl (1990): Psychoacoustics - Facts and Models, Springer-Verlag, Berlin.
[3] International Electrotechnical Commision (1995): IEC 268-18, Peak programme level meters - Digital
audio peak level meter, First edition.
[4] DK Audio MSD600C oversampling meter (1999), http://www.dk-audio.dk.
[5] M. Ankerman et al. (1992): Aussteuerungsmesser mit Anzeige der Kurzzeit-Abtastwerte-Verteilung,
ITG-Fachbericht 118, VDE-Verlag GmbH, p. 163-169.
[6] Internal report and Test CD, “Full Scale Reproduction Test no 1”, TC Electronic A/S.
Sine, Square and MLS signals at various levels.

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