Loudness standards in broadcasting.

Case
study of EBU R-128 implementation at
SWR
Carbonell Tena, Damià

Curs 2015-2016

Director: Enric Giné Guix

GRAU EN ENGINYERIA DE SISTEMES AUDIOVISUALS

Tr e ball d e F i d e G rau
Loudness standards in broadcasting. Case study of
EBU R-128 implementation at SWR

Damià Carbonell Tena

TREBALL FI DE GRAU
ENGINYERIA DE SISTEMES AUDIOVISUALS
ESCOLA SUPERIOR POLITÈCNICA UPF
2016

DIRECTOR DEL TREBALL

ENRIC GINÉ GUIX
Dedication

Für die Familie Schaupp. Mit euch fühle ich mich wie zuhause und ich weiß dass ich
eine zweite Familie in Deutschland für immer haben werde. Ohne euch würde diese
Arbeit nicht möglich gewesen sein. Vielen Dank!

iv
Thanks
I would like to thank the SWR for being so comprehensive with me and for letting me
have this wonderful experience with them. Also for all the help, experience and time
given to me. Thanks to all the engineers and technicians in the house, Jürgen Schwarz,
Armin Büchele, Reiner Liebrecht, Katrin Koners, Oliver Seiler, Frauke von Mueller-
Rick, Patrick Kirsammer, Christian Eickhoff, Detlef Büttner, Andreas Lemke, Klaus
Nowacki and Jochen Reß that helped and advised me and a special thanks to Manfred
Schwegler who was always ready to help me and to Dieter Gehrlicher for his
comprehension.
Also to my teacher and adviser Enric Giné for his patience and dedication and to the
team of the Secretaria ESUP that answered all the questions asked during the process.
Of course to my Catalan and German families for the moral (and economical) support
and to Ema Madeira for all the corrections, revisions and love given during my stay far
from home.

v
Abstract

During the 90s and the first decade of the 2000s the loudness levels of popular music
increased drastically as no real loudness standard was used for mixing and mastering.
The same happened in the broadcasting industry where constant loudness level changes
between programs or channels annoyed the consumers. Different organizations around
the world decided to stop this phenomena by creating standards like ATSC A/85 (USA,
Canada) or EBU R128 (Europe). A public German broadcaster, the SWR
(Südwestrundfunk), in which I had the opportunity to do my praxis semester, has
already implemented the EBU R128, and it is currently fully functional. I studied and
explained the European standardizations in order to increase the awareness about the
topic and to provide useful information for their fully understanding. In this work I
explain how the implementation was done at SWR, along with the insights from the
engineers who work there, providing a review with useful information for other
broadcasters.

Resum

Durant la època dels 90 i la primera dècada dels 2000, el nivell de loudness de la musica
pop va augmentar dràsticament ja que en aquella època no es feia servir cap estàndard
de loudness a l’hora de mesclar o masteritzar. El mateix va passar a la industria de les
telecomunicacions, on els canvis constants de nivell de loudness entre canals o
programes molestaven als consumidors. Diverses organitzacions d’arreu del món van
decidir acabar amb aquest fenomen creant estàndards com la ATSC A/85 (EEUU,
Canada) o la EBU R128 (Europa). Un canal públic alemany, la SWR
(Südwestrundfunk), on vaig tenir la oportunitat de fer el meu semestre de pràctiques, ja
ha implementat la EBU R128 i actualment està en complet funcionament. He estudiat i
explicat les estandarditzacions europees per augmentar la coneixença d’aquest tema i
per a proveir informació útil per la seva plena comprensió. En aquest treball explico
com es va realitzar la implementació a la SWR, conjuntament amb el parer dels
enginyers que hi treballen, proveïen així informació rellevant per a altres televisions.

vi
Prologue
With the digital revolution we achieved a new era for the audiovisual world and acquired
a technology that opened new opportunities. These new technologies improved the
dynamic range capabilities and reduced ground noise and the physical constrains inherent
to analog recording. It seems, however, that we were not able to fully understand the
advantages of this new technologies. The overwhelming increase in dynamic range
capabilities were despised by producing louder and increasingly over-compressed
material in order to sound louder than the rest, in an attempt to stand out (or perhaps, to
sell) more than our competition.

Even though compression is needed and in some situations -especially when audio is
meant to be reproduced in noisy environments such as cars or public areas- it has been
quite often over-used with commercial intentions. This situation, which began at the 60s,
speeded up at the end of the 1980s and reached its peak at the beginning of the 2000s, is
usually called “Loudness War”1, and it has reached a point where the product that
producers are selling is almost harmful for the consumers. (Vickers, 2010)

Lately there has been a lot of debate about the future of the audiovisual industry. So it is
that the International Associations in Telecommunications and Broadcasting has taken
measures so that modern technology can be exploited while protecting the consumer, who
has been complaining about the situation for years. Among other organizations, the
International Telecommunications Union (ITU) has worked on the ITU BS. 1770-2
standard and the European Broadcast Union (EBU) has worked on the EBU R128
recommendations. The aim of both is to set some ground rules for mixing in order to
normalize loudness levels and to avoid big changes in the loudness between TV programs
or channels, but these rules could be applied to the music industry as well. The aim of
this thesis is to study the implementation of these recommendations on a TV broadcaster,
and what changes have to be done in the workflow so these recommendations can be
fulfilled. The specific broadcaster to serve as a case study will be the
Südwestrundfunk2(SWR), a public German regional TV studio in Stuttgart.

This situation has been a problem in the music and in the broadcasting industry, but, as
said, in this thesis we will only be focusing on the broadcasting industry. Nevertheless, it
is going to be impossible not to mention the music industry constantly as they have
evolved together, and one cannot exist without the other. (E.g. Music produced for
Compact Disc often is also used in TV-Productions)

1
It is called Loudness war (also known as Loudness race) as there has been a tough competition to get the
mixings louder than the competitors and this, according to some experts (Shepherd., 2011), has been
harmful to the public audience, therefore it is compared to a war as both are competitive and harmful.
2
http://www.swr.de/

viii
ix
Content

Abstract ............................................................................................................................ vi
Prologue ......................................................................................................................... viii
Content ............................................................................................................................. x
Figures ............................................................................................................................. xi
Tables ............................................................................................................................. xii
1. HISTORY OF LOUDNESS...................................................................................... 1
1.1. Getting technical ................................................................................................ 2
1.1.1. Peak vs. loudness normalization................................................................. 4
1.1.2. Compression distortion ............................................................................... 5
1.2. Loudness in broadcasting................................................................................. 10
1.3. Metering ........................................................................................................... 10
1.3.1. Peak Program Meter ................................................................................. 11
1.3.2. The VU meter ........................................................................................... 11
1.3.3. New meters ............................................................................................... 12
2. RECOMMENDATIONS ........................................................................................ 15
2.1. ITU BS. 1770 ................................................................................................... 15
2.2. EBU R128 ........................................................................................................ 20
2.2.1. Program Loudness .................................................................................... 20
2.2.2. Loudness Range ........................................................................................ 20
2.2.3. True Peak level ......................................................................................... 22
2.2.4. Implementation in the production chain ................................................... 22
2.2.5. Loudness Parameters for short-form Content ........................................... 28
2.3. Other standards ................................................................................................ 29
3. STUDIOS SWR ...................................................................................................... 30
3.1. Structure ........................................................................................................... 30
3.2. Implementation of the Loudness regulations in the SWR ............................... 31
3.2.1. Loudness control equipment..................................................................... 32
3.2.2. Loudness in to the workflow .................................................................... 33
3.3. My experience with loudness .......................................................................... 34
3.3.1. Loudness in live broadcasting .................................................................. 34
3.3.2. Loudness in sound production and post-production ................................. 38
3.4. Daily problems, abuses and shortcomings ....................................................... 41
3.4.1. Automatic loudness level systems ............................................................ 42
3.4.2. Tricky situations ....................................................................................... 44
3.4.3. Commercial loudness abuse ..................................................................... 47
4. CONCLUSION ....................................................................................................... 48
5. TERMINOLOGY.................................................................................................... 50
6. BIBLIOGRAPHY ................................................................................................... 51

x
Figures
Figure i. Different track values according to the year of its release (Deruty, 2011) ........ 3
Figure ii. Effects of peak normalization (Katz, 2013, p. 67) ............................................ 4
Figure iii. Peak vs. Loudness normalization (EBU, 2011, p. 16) ..................................... 5
Figure iv. Compression cause distortion to the signal (Katz, 2013, p. 56) ...................... 5
Figure v. Frequency domain changes due to clipping (Katz, 2013, p. 61) ....................... 6
Figure vi. Amplitude increasing and clipping at amplitude 1 .......................................... 7
Figure vii. 440 Hz wave spectrogram ............................................................................... 7
Figure viii. Clipped 440 Hz wave spectrogram ................................................................ 8
Figure ix. 22533 Hz wave spectrogram ............................................................................ 8
Figure x. Clipped 22533 Hz wave spectrogram ............................................................... 9
Figure xi. Aliased harmonics from a 21533 Hz signal ................................................... 10
Figure xii. K-System scales (Katz, 2000, p. 8) ............................................................... 13
Figure xiii. ITU BS.1770 block diagram (ITU, 2015, p. 3)............................................ 16
Figure xiv. K-weighting curve........................................................................................ 16
Figure xv. Signal over-sampling for True Peak detection (Fleischhacker, 2014) .......... 19
Figure xvi. True-peak detection block diagram.............................................................. 19
Figure xvii. Loudness distribution, with gating thresholds and Loudness Range for the
film “The Matrix” (EBU, 2016a) ............................................................................ 21
Figure xviii. Different examples for Loudness Range depending on the replay
environment (EBU, 2011, p. 19) ............................................................................. 22
Figure xix. Scales of an "EBU mode" meter in LU (EBU, 2011, p. 24) ........................ 24
Figure xx. ARD structure ............................................................................................... 31
Figure xxi. Main display of a Lawo mc2 66 from SWR Stuttgart studios with loudness
meter in LU ............................................................................................................. 36
Figure xxii. Parameters in the Lawo console meters ...................................................... 36
Figure xxiii. External computer for Lawo mixing console monitor ............................... 37
Figure xxiv. RTW TM7 from Lawo mixing console ..................................................... 37
Figure xxv. RTW TM 9 touch monitor from a sound editing room in the SWR Stuttgart
studios...................................................................................................................... 39
Figure xxvi. Nugen Audio LM-Correct plug-in interface (NUGEN Audio, 2016) ....... 41
Figure xxvii. Graphic representation of a loud-soft file transition ................................. 42
Figure xxviii. Graphic representation of a soft-loud file transition ................................ 43
Figure xxix. Measurements from actual loudness transitions during broadcasting ....... 43
Figure xxx. Program scheme with -23LUFS voice level ............................................... 45
Figure xxxi. Program scheme with -23LUFS program level ......................................... 45
Figure xxxii. Voice level differences between programs ............................................... 46
Figure xxxiii. Loudness level difference between a film and a commercial .................. 47

xi
Tables
Table a. Aliased frequencies from 21533 Hz harmonics.................................................. 9
Table b. Summary of new Loudness units ..................................................................... 14
Table c. Weights for every audio channel of a 5.1 surround system .............................. 17
Table d. Listening levels and alignment signals summary ............................................. 27
Table e. Summary of the Loudness Parameters for short-form content (EBU, 2016b, p.
4).............................................................................................................................. 28
Table f. Regional broadcasting stations members of the ARD ...................................... 30

xii
1. HISTORY OF LOUDNESS

The history of music is far too long to take all into account, and most of the information
would be irrelevant for this case study. Therefore, we are going to start focusing on the
1940s.

In the 1940s, jukeboxes started to be popular in clubs and bars where music could be
played at any time, and it could be selected by the customers. At this point music started
to be produced for the mases and the business started to grow. Those jukeboxes were pre-
fixed to a loudness level by the owner of the club or bar in question, so the audience could
hear the music properly.

Also in the 40s, the VU meter had just been born in the broadcasting industry, but it was
then also used in the music industry. This was the first meter that intended to represent
the way humans perceive loudness, instead of representing only the signal characteristics.
(Audio Engineering Society, 2014)

Later on, during the 50s and 60s, 45 rpm singles appeared and producers started to figure
out that louder mixes were more frequently played in the jukeboxes, and they outsold the
others. Therefore, producers started to mix louder discs. (Sreedhar, 2007). Also, radio
broadcasters started to contribute to the war with the top 40 list. Producers started to mix
hotter singles in order to get into the top 40 list. As always, in a “louder-is-better”
scenario, the louder it sounded, the higher the probability you had to get into the list. Also,
advertising started to be an important factor to the albums sales. (Devine, 2013)

During the 60s and 70s, artists used to make compilation albums with all their hits. At
this time, when songs from different years were put together, they realized that older
songs sounded softer than newer ones, so the older songs had to be remastered to make
them sound louder.

That was the beginning of what we call the Loudness War. But, because of the physical
constrains of the physical support for the recording at that time, the recordings could not
reach the level of compression that they have nowadays. This is because at that time vinyl
used to be the main carrier, and records were transferred as an analogue electroacoustic
wave to the vinyl disc. This wave had to be readable with a needle, and needles cannot
follow discontinuities in a wave. Such limitation would impose a typical crest factor3 (CF)
of 14dB.

With the digital revolutions most physical constraints were gone, and with the CD we
would be able to theoretically record any kind of wave, up to a CF = 0dB (such as a square
wave). Quantization distortion could not be avoided, but with 16 bits per sample and the
clever use of dithering (turning quantization THD into white noise), the Signal to Noise
Ratio (SNR) had never been bigger.

It is at this point where producers and musicians, and also broadcasters, started to explore
the possibilities of the digital era. At first, the dynamic range [1] was being used to create
impact with loud and soft sounds, but as what happened with single discs in the 50s, music
started to become louder (and thus dynamics decreased) in order to sell more. Digital

3
the crest factor (CF) is the ratio, usually expressed in dBs, between a maximum peak value and the
effective root-mean-square average value of an alternate signal such an audio signal

1
compressors and peak limiters where much more powerful than analog ones, so music
was compressed beyond expectations.

This compression reached its crest around 2008, when the album Death Magnetic from
Metallica was released (Shepherd, So, Justin Bieber is louder than Motorhead, AC/DC
and The Sex Pistols… – wait, WHAT ?, 2011). It is one of the loudest albums ever made,
and it was so compressed that customers rapidly reported the bad sound quality of the
record.

However, music industry has lately made a huge change with the arrival of new cloud-
based playback technologies such as iTunes or Spotify. These platforms changed the way
people listen music and lately applied also a loudness normalization per track by default.
That means that they match a uniform loudness level for all their content, so people do
not have to adjust levels when listening to different tracks. That means that highly
compressed works would be punished as they would sound softer and with less dynamics
than the more dynamic mixes. This can be a significant step to end the Loudness War as
these platforms are the main music distributors nowadays.

Also, lots of influent music producers and big music technology companies are pursuing
to make a change in the industry to get dynamics back, accepting and following the new
recommendations for loudness normalization.

1.1. Getting technical

To demonstrate all that has been explained we need some empirical facts. In 2011, the
magazine Sound on Sound released an article4 where the RMS value of 4500 hits from
1969 until 2010 was compared and the results plotted in a timeline. The outcome was
quite impressive as we can see clearly the loudness increase.

But that was not the only consequence of the increased loudness values: the corresponding
reduction in dynamics (see CF) and also the increased percentage of peak overs
(instantaneous values above -1dBFS) can also be seen.

4
see Dynamic range and the loudness war – SoS, Sept. 2011 -
http://www.soundonsound.com/sos/sep11/articles/loudness.htm

2
Figure i. Different track values according to the year of its release (Deruty, 2011)

3
 1.1.1. Peak vs. loudness normalization

To achieve these high RMS values, we need to compress the signal. This compression is
needed as nowadays, it is a common practice to do peak normalization. Peak
normalization consists in adjusting the level of a piece by using break wall limiters until
the highest peaks hit full scale. So, compression is used to reduce the distance between
peaks and the average level of loudness of the piece, driving CF to figures no higher than
3-6dB. By doing it, we can get the average loudness level much higher.

Peak normalization has been one of the main

contributors to the Loudness War, as pieces with high
peaks such as percussive ones will get a lower average
loudness level than the ones with low peaks. That can
cause, as Bob Katz explains in his book “iTunes music”,
that a string quartet would sound louder than a
symphony orchestra as the orchestra has much higher
crest factor [2]. As shown in the figure ii, the string
quartet has a higher average level when normalized.

In order to avoid that, producers use to compress or peak

limit their works so they can, even though it sounds
crazy, make an orchestra sound louder than a string
Figure ii. Effects of peak normalization
quartet.
(Katz, 2013, p. 67)
This practice became more and more usual and that is the
reason we can see a clear increase of the RMS value of
main hits during the last decades, even though there seems to be a turning back with the
new ways of music distribution.

This same trick has been used in TV. Commercials are over compressed in order to have
a much higher RMS value than a film or a program so it will sound much louder. The
more compressed, the louder your commercial will sound compared with your
competitors’. As shown in the figure below, peak normalization encourages compression
in order to sound louder than other competitors. However, loudness normalization, as we
will show in further sections may be the solution for loudness changes between contents.

4
 Figure iii. Peak vs. Loudness normalization (EBU, 2011, p. 16)

1.1.2. Compression distortion

Compression has always been present in the history of music and broadcasting, and it can
be used as an artistic resource, but over-compression can cause distortion that could
decrease the sound quality. Compression and distortion are highly related and when we
have compression we would inevitably have distortion too. We can see the effects of
compression in the next figure:

Figure iv. Compression cause distortion to the signal (Katz, 2013, p. 56)

This distorted wave sounds much louder as we have increased the average level, and also
because, when distortion occurs it usually generates high frequency content and this is
perceived louder than low frequency content.

5
This frequency content alteration can also appear by clipping the signal. Clipping means
increasing the level of a signal until it loses the curvy form of a wave and it becomes
squared. That happens because we are trying to increase the level beyond the maximum
peak level, and the wave has to be clipped. By clipping we do not increase the sample
peak level [3], but we do increase the average level and the true peak level [4]
(intersample peak) [5].

Clipping does change the wave form and, therefore, it changes the frequency content of
the signal. In an analog system it can be desired to enrich the sound as it would produce
harmonic distortion. However, in a digital domain, the distortion caused is not completely
harmonically related to the original wave, but it is partially caused by aliased harmonics.
It should be kept in mind that digital clipping in a post-production context would appear
after sampling, and thus, after the anti-aliasing stage. In the next figure, we can see the
effects of very high frequency clipping in the frequency domain:

Figure v. Frequency domain changes due to clipping (Katz, 2013, p. 61)

This is an image of a 21.533 kHz signal with a sampling rate of 44100 samples per second.
This frequency can almost not be heard, but the distortions appeared due clipping are
completely audible. Those new frequency components are harmonics of the aliased
frequencies created by clipping.

We have our own tests in order to confirm Katz results. To test the distortion due to
clipping, we took a 440 Hz wave and we periodically increased its amplitude and clipped
it to get a clipped wave of amplitude 1. The wave form can be seen in the next figure:

6
 Figure vi. Amplitude increasing and clipping at amplitude 1

If we analyze the first part of the wave we would get a really defined spectrum with big
peak at 440 Hz. However, if we analyze the second part (strongly clipped), we would
obtain a much more different spectrum with a lot of different frequency components. We
can see though that all the new frequencies are higher than the fundamental frequency.
That is because to square a wave we need to add the odd harmonics form the fundamental
frequency to the original signal. As we can appreciate in the Figure viii the new peaks
that appear in our spectrogram correspond to 1320 Hz, 2200 Hz, 3080 Hz, etc., which are
in fact the odd harmonics of 440 Hz.

Figure vii. 440 Hz wave spectrogram

7
 Figure viii. Clipped 440 Hz wave spectrogram

But what would happen if the fundamental frequency chosen is really high? The
harmonics from the fundamental frequency would be higher than the Nyquist frequency
(22050 Hz in this case) and even higher that the sampling frequency, therefore, they will
produce aliasing to our signal. To see the effects of this distortion we reproduced the
experiment as before, and we took the exact same frequency from the Katz experience,
21533 Hz. We clipped at amplitude 1 and we have generated the spectrums from the pure
wave, and from the clipped wave.

Figure ix. 22533 Hz wave spectrogram

8
 Figure x. Clipped 22533 Hz wave spectrogram

We can see that the effect is similar to the previous experiment with the 440 Hz, but this
time the new frequency components that appear in out spectrum are below the
fundamental frequency, as they are aliased harmonics from the fundamental frequency,
21533 Hz.

If we calculate the odd harmonics of our signal we will get really high frequencies (for
an audio signal). The three first odd harmonics from 21533 Hz are: 64599 Hz, 107665 Hz
and 150731 Hz. To calculate the aliased frequency from those harmonics, we just have to
subtract the value of the nearest multiple of the sampling frequency, which are 44100 Hz,
88200 Hz and 176400 Hz respectively. The results of this process are the next
frequencies: 20499 Hz, 19465 Hz and 25569 Hz. The two first are lower than the Nyquist
frequency, but the last one still higher, so it is going to produce aliasing again. By doing
the same process we obtain that the aliased frequency is 18531 Hz.

Harmonics from 21533 Hz Closest sampling frequency Aliased frequency

(𝒇) multiple (𝒇𝒔 ) 𝒇𝒂 = |𝒇 − 𝒇𝒔 |
64599 Hz 44100 Hz 20499 Hz
107665 Hz 88200 Hz 19465 Hz
150731 Hz 176400 Hz 25569 Hz → 18531 Hz
Table a. Aliased frequencies from 21533 Hz harmonics

If we look carefully at the spectrum of the clipped signal we will see that the peaks
appeared correspond with those frequencies.

9
 Figure xi. Aliased harmonics from a 21533 Hz signal

All those artifacts are in fact happening nowadays with over-compressed, post-produced
audio materials. We are getting used to the way it sounds, and it may sound natural for
some people5. But it is not, and expert listeners would notice them. These are also reasons
why regulation and standardization are needed.

1.2. Loudness in broadcasting

Broadcasting has also been affected by the loudness war. It is clear that radio broadcasters
have been affected as they use the material that producers and musicians are making, and
this material has become louder in the past decades, even though radio stations have also
feed backed this war.

Despite that, loudness war is also present in other aspects in broadcasting. Loudness level
differences between programs or between channels is very frequent, and very loud
intervening commercials are the main source of complains, as this has become more and
more usual lately. Again, the louder, the more it will stand out and eventually sell out.

1.3. Metering
Loudness metering has always been an important issue. That is why there are many
different types of meters and ballistics. There are meters for many different characteristics
of a signal, but we are going to focus on those that have more importance in the loudness
area.

5
See Loudness Normalization: Paradigm Shift or Placebo for the Use of Hyper-Compression in Pop
Music? - http://quod.lib.umich.edu/i/icmc/bbp2372.2014.143/1

10
The Peak Program Meter (PPM) measures the highest sample peaks of a signal (even
though there are several meter variations). With this measure we can know the highest
peak from our signal, but this does not give a good representation of its loudness.

The VU meter, on the other hand, was created to obtain a more realistic representation of
loudness by giving out an averaged value of the signal. Even though it has been in use for
many years, there was a need to create new meters to really represent how humans
perceive loudness, and also to represent other characteristics of a sound like the Loudness
Range.

1.3.1. Peak Program Meter

The peak program meter (PPM) is a measuring instrument to measure the level of an
audio signal which was first introduced in the professional fields during the early 1930s
(Yonge, 2008). It usually consisted on a needle moved mechanically by electromagnetic
impulses created by the analogue audio signal. It is nowadays used for digital audio
signals, typically as a bar graph made of a light array displayed vertically or horizontally.
This meter has had many different variations as well as many different scales.

Firstly, the basic PPM is the True Peak meter, which indicates the peak value no matter
the duration of the peak.

Secondly, the Quasi Peak meter only indicates the peaks with a certain minimum duration
of a few milliseconds, about 10 ms (Schmid, 1976). The peak duration time needed is
determined by the integration time, so shorter peaks will not have enough weight and
their true peak level will not be shown.

Another approach is the Sample Peak meter, which shows only the sample peaks of a
digital audio signal, but not the true peak that may be between two samples. Although it
may have an integration function for true peaks, another solution to that is the Over-
Sample Peak meter, which first oversamples the signal, and then displays the sample peak
of the oversampled signal.

Since it is an old technology, there have been many different approaches of it. Many
important broadcasters associations have made their own scale for their convenience, and
as a result there are a large number of scales in use. Engineers needed a new, trustable
and standard technology to work with. Also, regarding loudness metering, the peak meter
does not give good information about how humans perceive the loudness of a signal. The
VU meter was then introduced.

1.3.2. The VU meter

The VU meter was firstly introduced in 1942 and it was first used in broadcasting stations.
It is one of the simplest meters for audio metering, and it has been in use since its creation
in the broadcasting and the music industry. It is also a measuring instrument that measures
the level of an audio signal, but it has a different approach than the peak meter. The VU
meter is intentionally slow, it does not react to rapid changes in the level of the audio
signal. Instead, it averages the signal within a rise/fall time of 300 ms, and gives a better

11
approach of how humans perceive loudness (Johansen, 2006). It could be understood as
an approximation of the root-mean-square (RMS) value of the signal.

The VU meter uses VUs (Volume Units) as a measurement unit, where 0 VU equals the
intensity of a 1KHz signal at a reference level that has been applied during 300 ms. The
VU meter has to be calibrated at a reference level, which it is usually 1.23 Vrms (+4 dBu
over a 600 Ohm resistance) for a pure tone (Schmid, 1976).

1.3.3. New meters

As seen with the previous technologies there was a need to create a new metering system
that could measure loudness better, and that could be used for all the parties interested.
There was a need for standard metrics that could measure loudness once and for all and
that could be used not only for all the different genres of music but also for all kinds of
broadcasters, film makers and producers.

1.3.3.1. K-System
Bob Katz, a prestigious mastering engineer, proposed this system to set some ground rules
for all situations6. He divided genres in three groups: those that do not need much dynamic
range, such as broadcasting and radio content or pop music; those that need slightly more
dynamic range, such as rock and country music and moderately-compressed content
intended for home listening; and finally, those that do need a lot of dynamic range, such
as classical music, hi-fi recordings and cinema. These three groups were divided in three
different set ups, K-12, K-14 and K-20, respectively.

They are named after the headroom [6] they each have, so K-12 has 12 dB of headroom
and so on. Due to these dynamic range changes, the studio monitor gain must be adjusted,
before mixing, depending on what we are mixing. To do it, we have to generate a standard
pink noise of -20dBFS, play it out loud through the studio monitors, and adjust the gain
until we measure 83dBSPL in the sweet spot using the C ponderation curve. At the end
of this process, we would have set our system to the K-20 standard. If we would like to
set our system to K-14 or K-12, we would have to low the monitor gain -6 and -8 dB
respectively.

6
see http://www.aes.org/technical/documentDownloads.cfm?docID=65

12
 Figure xii. K-System scales (Katz, 2000, p. 8)

With this ground rules set, all producers could mix at the same level and, by introducing
the information of the set up used in the metadata of the file, broadcasters or customers
could reproduce the piece at the right loudness level, adjusting it for every file in order to
get a constant loudness level.

1.3.3.2. LUFS, LKFS, LRA

Even though Katz tentative did not get much practical success at first, it fostered the
discussion that led international organizations in telecommunications and broadcasting
start developing new standards, which we will deeply discuss later on. With those
standards came new metrics again, the LUFS and the LKFS. Both are the same unit but
with different names as they were proposed by different organizations.7

In the BS.1770, the ITU proposed the LKFS as an absolute loudness unit. It is a Loudness,
K-Weighted, referenced to digital Full Scale (LKFS) unit intended to measure the average
perceived loudness of a piece and give a value for the whole audio content. That means
that it is a psychoacoustic loudness unit, as the K-weighting curve represents how humans
perceive sounds.
The EBU proposed different units, the LUFS (Loudness Units, relative to digital Full
Scale) as the same concept and functioning as the LKFS. Also, the LU (Loudness Units)
which is a loudness unit relative to a target level, which we will discuss later on. And

7
They were not the same, at first, when LKFS were presented in the ITU-R BS. 1770 and they were not
equivalent until the publication of the ITU-R BS. 1770-2, where the ITU included some rectification
according to the EBU R 128 recommendations.

13
finally, the LRA (Loudness Range) is meant to represent the dynamic range of a whole
program. It was first suggested by TC-Electronics and finally added to the EBU R128.

As said before, 1 LKFS equals 1 LUFS, and they are both equivalent to 1 dB. The LU is
a relative unit which describes loudness level differences and it is also equivalent with
the dB scale. LRA, on the other hand, is an absolute unit that describes the overall
program loudness range, and it is measured in LU. In other words, it measures the number
of LUs between the softest and the loudest part of the piece. However, it ignores extreme
events in order that they do not affect too much to the overall measurement.

LU Loudness Unit
LKFS Loudness, K-weighting, with reference to Full Scale
LUFS K-weighted Loudness Unit with reference to digital Full Scale
LRA Loudness Range, measured in LU
Table b. Summary of new Loudness units

These meters have another characteristic, they are gated. That means that they measure
the average loudness of a whole program or piece of audio, but they omit the softer parts
below a certain threshold, which will be discussed later on. This is because humans
identify how loud a piece of audio is, because of the loud parts but not for the soft ones.
Moreover, these new meters are meant to measure loudness for all genres, from movies,
to ads, to classical pieces, and pop songs. Gating allows these new meters to work no
matter what they are measuring, and they are meant to make possible to match the
loudness level of a 2 hours film with a 20 seconds commercial.

14
 2. RECOMMENDATIONS

2.1. ITU BS. 1770

The Radiocommunication Sector from the ITU issued ITU BS.1770, a paper in the
Broadcasting Service (sound) series, with recommendations of audio measurement
algorithms to determine an objective approximation of the subjective loudness level of a
program together with a true-peak measurement approach. It is one of the most important
standards, as it is used by several other standards from different broadcasting unions.

There have been several versions of this recommendation in which each version has
included new characteristics or features in order to improve the algorithm. As the ITU
has included in the recommends section of the paper, it is intended to update the
recommendation when new algorithms had been developed. (ITU, 2015)

The ITU considered several aspects to build the algorithm, such as:

- the wide dynamic range that new technologies offer

- the fact that most of the productions are a mix of mono, stereo and multichannel
signals
- the fact that listeners desire a uniform subjective loudness level

With all that in mind, they built the algorithm and recommended it to be used when an
objective measurement of the loudness level of a multichannel audio is needed. Thus,
indicators of loudness levels used in production and post-production should be based on
these recommendations. (ITU, 2015)

As said before, the aim of this algorithm is to obtain an objective approximation of the
subjective loudness level from a whole audio file. Therefore, this algorithm must take
into account psychoacoustic concepts. To understand the implications and the
implementation of the algorithm we will explain it step by step. However, in this thesis
we will not review the background and the methodology used to develop the algorithm.
To know more about this you can check the ITU BS.1770-4 quoted in the bibliography.

The multichannel algorithm is based on a previous one from a study (Soulodre, 2004) to
obtain a loudness indicator for mono signals. This algorithm is known as Leq(RLB)[7].
It was designed to be very simple, and it is based on a high pass filter, known as the
revised low-frequency B-curve (RLB), which is a modification of the weighting B-curve,
followed by a root mean square of the sound level from a given time period, Leq
(equivalent continuous sound level). After several subjective tests it was concluded that,
despite its simplicity, this algorithm performed very well for monophonic signals.

Implementing a multichannel algorithm presents several more challenges than the

monophonic one, as it has to work for mono, stereo and multichannel signals. With regard
to the good performance of the previous algorithm, the multichannel one is based on the
monophonic sound level measurement algorithm.

Now we will study the multichannel algorithm, and to begin with, a block diagram will
help us to understand the algorithm as a whole. As we can see in figure xiii, the algorithm
is thought as a multichannel algorithm based in a 5.1 system, but it does not take into
account the low frequency channel to measure the loudness level. We can also see that

15
the first step is to apply a K-filter [8], as this will give a psycho-acoustics based approach
of the sound that compensates the acoustics effects of the head and takes into account
how differently frequencies are perceived. Then, the mean square value of the signal is
calculated and the result is multiplied by a factor. This factor is different depending on
the channel. The surround channels are increased to compensate the perceived gain of
those channels because of their position on each side of the listener (Qualis Audio, Inc.,
2013). Finally all signals are summed up and gated. This gating was firstly introduced in
the third version of the recommendation.

Figure xiii. ITU BS.1770 block diagram (ITU, 2015, p. 3)

The K-filter is used to compensate the acoustic effect of our head and it is implemented
in two steps. First, a shelving filter and then a simple high pass filter is applied. This high
pass filter is known as the RLB weighting. The filter coefficients of the filters applied
would change depending on the sampling rate of the signal.

Figure xiv. K-weighting curve

16
Once the signal has been filtered, the mean square of the signal is calculated. It is
calculated in intervals of length 𝑇:
𝑇
1
𝑧𝑖 = ∫ 𝑦𝑖2 d𝑡
𝑇
0

Where 𝑧𝑖 is the mean square value, 𝑦𝑖 is the filtered input signal, and 𝑖 represents each
channel.

Then, the signal loudness for an interval 𝑇 is expressed as:

𝐿𝑘 = −0.691 + 10 log10 ∑ 𝐺𝑖 ∙ 𝑧𝑖 𝐿𝐾𝐹𝑆

𝑖

This value is expressed in LKFS as it is a K-weighted signal, and as we can see in the
formula, it is in a logarithmic scale. The -0.691 value is a constant value to calibrate the
effects of the two filters (K-filter and RLB), as they modify the gain of the signal. (Carroll,
Jones, & Williams, 2007). And, 𝐺𝑖 is the weighting factor for every channel.

Channel Weighting 𝑮𝒊
Left (𝐺𝐿 ) 1.0 (0 dB)
Right (𝐺𝑅 ) 1.0 (0 dB)
Centre (𝐺𝐶 ) 1.0 (0 dB)
Left surround (𝐺𝐿𝑠 ) 1.41 (~+1.5 dB)
Right surround (𝐺𝑅𝑠 ) 1.41 (~+1.5 dB)
Table c. Weights for every audio channel of a 5.1 surround system

Once the loudness calculation process is understood, we need to know how the gated
loudness level is measured. The gating function is needed because not all intensities
contribute the same way to the overall perceived loudness level, as the loudness level of
a signal is described mostly because of the loud parts than the soft ones.

The calculation of the gated loudness level is done by blocks. The previous signal of
length 𝑇 is divided in blocks of 𝑇𝑔 = 400 𝑚𝑠 which are overlapped a 75% with each
other. The gating loudness measurements are performed coinciding with the blocks
length, but those blocks that are incomplete at the end of the signal are left out.
The mean square value of the different blocks is calculated as:

𝑇𝑔 ∙(𝑗∙𝑠𝑡𝑒𝑝+1)
1
𝑧𝑖𝑗 = ∫ 𝑦𝑖2 𝑑𝑡
𝑇
𝑇𝑔 ∙𝑗∙𝑠𝑡𝑒𝑝

Where 𝑠𝑡𝑒𝑝 is 1 − 𝑜𝑣𝑒𝑟𝑙𝑎𝑝, 𝑗 is the gating block and 𝑖 represents every channel.
So, as before, the loudness level of every block is calculated as:

17
 𝑙𝑗 = −0.691 + 10 log10 ∑ 𝐺𝑖 ∙ 𝑧𝑖𝑗
𝑖

If we set a loudness threshold Γ, we will obtain a set of gating blocks that its loudness
level is above the threshold, this subgroup is 𝐽𝑔 = {𝑗: 𝑙𝑗 > 𝛤}, and its number of elements
is |𝐽𝑔 |. The number of elements in this subgroup depends on the threshold fixed. The
gating process is made in two steps, first with an absolute threshold Γ𝑎 and a relative
threshold Γ𝑟 . The absolute threshold is fixed at Γ𝑎 = −70 𝐿𝐾𝐹𝑆, and the relative
threshold is calculated by subtracting 10 to the loudness value obtained applying the
absolute threshold. So the first gated loudness level from the interval 𝑇 is calculated as:

1
𝐿𝐾𝐺1 = −0.691 + 10 log10 ∑ 𝐺𝑖 ∙ ( ∙ ∑ 𝑧𝑖𝑗 ) 𝐿𝐾𝐹𝑆
|𝐽𝑔 |
𝑖 𝐽𝑔

Where:

𝐽𝑔 = {𝑗: 𝑙𝑗 > Γ𝑎 }
Γ𝑎 = −70 𝐿𝐾𝐹𝑆

Then the relative threshold is calculated as:

Γ𝑟 = 𝐿𝐾𝐺1 − 10 𝐿𝐾𝐹𝑆

And the final gated loudness level would be:

1
𝐿𝐾𝐺 = −0.691 + 10 log10 ∑ 𝐺𝑖 ∙ ( ∙ ∑ 𝑧𝑖𝑗 ) 𝐿𝐾𝐹𝑆
|𝐽𝑔 |
𝑖 𝐽𝑔

Where now:

𝐽𝑔 = {𝑗: 𝑙𝑗 > Γ𝑟 }

This gating was firstly introduced in the third version of the recommendation in regard of
the later EBU R128 recommendations.

Another issue covered in this recommendation is the peak detection. The ITU stated that
true peak meters should be used for peak detection. Nowadays, in broadcasting, quasi-
peak or sample peak detector meters are normally used to detect peaks. They are called
quasi-peak detectors as they have a reaction time of approximately 10ms. This means that
smaller peaks are not properly detected. Furthermore, peaks may not always be
represented by a sample, as there can be inter-sample peaks. Initially, those peaks could
not be detected and therefore the headroom needed could not be properly calculated,
eventually causing distortion.

To solve those problems, the ITU recommends the use of an algorithm for accurate true-
peak detection. This algorithm is based on some simple steps:

18
Firstly, the signal has to be attenuated 12.04 dB (that is, to ¼ of signal voltage) to leave
enough headroom for the next steps of the process.

Then an over-sampling process is performed. The ITU assumes a 48 kHz sampling rate,
and therefore, recommends a 4 times over-sampling process in order to achieve a
sampling rate of 192 kHz. With this, we can obtain a more accurate representation of the
waveform in order to represent the peaks with samples.

Figure xv. Signal over-sampling for True Peak detection (Fleischhacker, 2014)

After that, the signal is filtered with a low-pass filter, and then the absolute value of the
samples is calculated by inverting the negative values. At this point we would have a
number with the true peak value of the signal. Nevertheless we still have to increase the
level of the signal by 12.04 dB to compensate for the previous attenuation. The result of
this process should be expressed as dBTP (decibel referenced to digital Full Scale
measured with a True Peak Meter (EBU, 2011)) once it is converted to a logarithmic
scale. Here we present a block diagram for the whole process.

Figure xvi. True-peak detection block diagram

19
 2.2. EBU R128
The EBU R128 recommendation has been developed by the PLOUD EBU research
group. It has extra material about Loudness Metering Specifications (EBU Tech 3341),
Loudness Range Descriptors (EBU Tech 3342), Loudness Production Guidelines (EBU
Tech 3343) and Distribution Guidelines (EBU Tech 3344). All together they establish a
well specified work-flow methodology in order to help professionals from the
broadcasting sector to identify and measure the loudness level for all contents.

The EBU R128 is based on the ITU BS.1770, but it extends its content by introducing
new concepts and defining some targets. In this recommendation, a Loudness Target
Level has been defined, together with a gating method for loudness normalization to
assure loudness matching between contents. In order to achieve this, three new key
concepts have been introduced, Program Loudness, Loudness Range and Maximum
Permitted True Peak Level.

2.2.1. Program Loudness

To understand program loudness, we need to define a program. As the EBU recommends,

in this document a program is understood as every single audiovisual content, no matter
it is a film, a show or a commercial. Knowing that, program loudness is defined as the
long-term integrated loudness over the duration of a program (EBU, 2011, p. 11). It is
expressed with a number (with one number after the decimal point) that indicates the
average loudness of a program in LUFS (or LKFS). This value is calculated following
the ITU BS.1770 methodology explained in the previous section, but including a gating
function.

The gating function basically excludes from the measurement all the parts from the
program that are softer than a certain threshold. After several series of listening tests, this
threshold was fixed at -8 LU taking as reference the loudness level of the ungated program
in LUFS. This gating function was not firstly introduced in the ITU BS.1770-0, but it was
in the ITU BS.1770-2 version, although the ITU considered a threshold of -10 LU instead.
After consideration, the EBU has accepted the proposal and the gating level is set at -10
LU since then as well.

The tests also showed which should be the target loudness level for all programs. The
target level should be -23.0 LUFS. However, it has been set an acceptance value of ±1
LU for technical difficulties or unpredictable programs such as live shows.

2.2.2. Loudness Range

Loudness Range (LRA) quantifies (in LU) the variation of the loudness measurement of
a program based on the statistical distribution of loudness within a program (EBU, 2011).
With the statistical distribution very loud isolated elements would not affect the overall
loudness measure, as extreme cases are excluded.

The calculation of the loudness range is made by taking a vector of loudness levels
obtained by using a 3 second overlapped sliding windows and a cascade gating method
with an absolute and relative gate, as the ITU BS.1770 specifies. With this gating system,

20
extreme soft events are eliminated from the measure thanks to the absolute gate, and it
makes the measure independent of the signal level thanks to the relative gate.

The gated loudness level values are distributed and the distribution width is quantified
using a percentile range. The LRA is defined as the difference between the 10th and the
95th percentiles. This way extreme events are eliminated of the measurement, because,
for example, a fade out at the end of a song or a single loud gunshot would deviate the
measure and increase the LRA drastically (EBU, 2016a).

Figure xvii. Loudness distribution, with gating thresholds and Loudness Range for the film “The Matrix” (EBU, 2016a)

A maximum Loudness Range is not defined in this recommendation as there is no

loudness range capable to fulfill all demands. Every genre should have its loudness range
because an action film cannot have the same loudness range as a news magazine. In
addition, we cannot define a loudness range for all listening conditions. As shown in the
next figure, every listening environment and amplifying system has its own needed
loudness range.

21
 Figure xviii. Different examples for Loudness Range depending on the replay environment (EBU, 2011, p. 19)

That is the reason why EBU encourages the use of Loudness Range as it is a clear
indicator of whether dynamics processing such as compression is needed. Also because
it can indicate if there has been a process in-between the production chain that has
changed the original dynamic range.

2.2.3. True Peak level

As explained before (3.1. ITU BS. 1770) inter-sample peaks can be an issue, as they
cannot be detected by the most common used sample-peak meters. Therefore, in the
R128, EBU encourages to follow the ITU RS.1770 recommendations to perform proper
true peak detection.

As peak normalization is left behind, now, with loudness normalization, peaks are still a
concern, as we must be careful with its levels. Therefore, the EBU recommendation also
sets a Maximum Permitted True Peak Level. They recommend -1 dBTP in order to avoid
distortion that may occur in further production chain stages. To be able to detect and treat
peaks, the new loudness meters should have the “EBU mode”, which includes the True
Peak meter, in all the stages of a production.

2.2.4. Implementation in the production chain

There are two ways to get the program to the target level. On the one hand, we can keep
the mixing habits that we already have and do a level shift afterwards. On the other hand,
we can change our mixing habit and focus the mixing towards the target level, so no level
shifting will be needed afterwards.

The first method is legitimate, and it can be useful in some occasions. During the
transition time (from peak to loudness normalization) can be helpful as engineers can get
used to the new levels comparing their mixings with the shifted ones. Also, in direct
programs where the target level could not have been achieved due to unexpected events,

22
this method could be really useful, although there is a ±1 LU of tolerance. It is worth
considering, that most of the times, the shifting that will have to be performed would be
negative (attenuation), and therefore, there would be no need to do any more calculations.
Although, if the shift has to be positive, we would have to check the dynamic range and
the maximum true peak level again.

Even though, the first method can be used, the second one is recommended. Changing
the mixing habits will be good, as engineers could stop worrying about the hitting the top,
and they could mix by ear once they would be used to it. This will result in much more
dynamic mixes, and as the Maximum True Peak Level is set at -1 dBTP, hitting the top
will then rarely be a concern.

To change the mixing habits and to be able to mix more freely and dynamically as before,
engineers need a new loudness metering system, that allows them to check the loudness
level at any time. Therefore, the EBU R128 also includes a new metering system.

2.2.4.1. EBU meters

This system is based in three different time scales, Momentary, Short-term and Integral
Loudness.

 The shortest time scaled is the Momentary Loudness scale (abbreviated as “M”).
It uses a sliding rectangular time window of 400 ms of length. The measures that
indicates are not gated.
 The second shortest is the Short-term Loudness scale (abbreviated as “S”). It also
uses a sliding rectangular time window, but this time it is 3 s long. Its measures
are also not gated.
 And finally, the longest scale is the Integrated Loudness scale (abbreviated as “I”).
It measures the average loudness value of the whole program, no matter its length.
This measure is gated as the ITU BR. 1770 recommends [2.1. ITU BS. 1770].
(EBU, 2016c)

The new meters should also show the Loudness Range and the True Peak Level of the
signal, as they are also very important parameters to follow the EBU recommendations.
PLOUD group [9], and several manufacturers have agreed to produce new meters that
include the “EBU mode” that should follow all the previous indications. All the
specifications about the new meters can be found in the EBU Tech 3341 document. (EBU,
2016c)

The EBU Tech 3341 also recommends two different types of meter scales, depending on
the dynamics required of every production and the comfort of the engineer. The two scales
are the “EBU +9 scale” and the “EBU +18 scale”. The first one may be used by default,
and it has a range of −18.0 LU to +9.0 LU (−41.0 LUFS to −14.0 LUFS), and the second
one has a range of −36.0 LU to +18.0 LU (−59.0 LUFS to −5.0 LUFS). (EBU, 2016c)
The meters can provide an absolute value expressed in LUFS or there can be fixed a 0
value (according with the recommendation at -23.0 LUFS=0.0 LU) so the measure would
be a relative value expressed in LU.

23
 Figure xix. Scales of an "EBU mode" meter in LU (EBU, 2011, p. 24)

2.2.4.2. What to measure

All the new meters created and explained before have to be used to measure the loudness
characteristics of a signal. What matters here is to define which signal to measure in order
to get representative information about the loudness level of a program.

The EBU R128 recommends measuring the entire program as it is a method that will
assure a correct measure for any case or genre. However, there is another option. It is also
possible to choose an anchor signal form the program (normally dialog) and determine its
level alone, adjusting all the other parts in reference to this one.

The anchor signal must be a central and important part of the sound scope. This method
can be useful in wide loudness range programs, but choosing the signal is a complex
process which requires experienced engineers, and it is only recommended once the
operators are fully familiar with the loudness normalization process. It is worth saying
though, that there are automatic anchor signal discriminator algorithms that may help in
this process, but they do not work perfectly for all situations.

As said, this method can be useful for wide LRA programs, but in narrow LRA programs,
such as commercials, the difference between the anchor signal and the whole program
level may be small. As the biggest common denominator, R 128 recommends to measure
the whole program with all its elements instead of anchors, even with wide LRA material.
(EBU, 2011)

24
 2.2.4.3. File Based System
Nowadays, most of the broadcasters have a file-based production workflow. In this
working scheme, all the previous loudness recommendations remain the same, loudness
level normalization and dynamic control should be done during the production of new
material. With the old production saved in the archive there are several options to
normalize the loudness level. All the next options are valid and the choice of any of them
depends on the company structure and workflows.

 Actually changing the loudness level of all the files from the archive material and
set it to the target level. This may sound tedious, but there are automatic or semi-
automatic hardware and software systems that have a good and relative fast
performance.

 Actually changing the loudness level only when needed. That means that when a
file is peaked from the archive, it will be normalize before is used or sent.

 Another option is not to modify the file, but measure its loudness level and adjust
the playout level before it is broadcasted, without changing the loudness level of
the file, in order to achieve the target level only when broadcasting. Of course all
the material has to be measured before the broadcasting.

 And lastly, using the correct metadata, the loudness level of the file can be
transmitted to the consumer reproduction system, and there the loudness level is
adjusted to achieve the target.

Independently of the choice, it is undeniable, that metadata can play a big role in a file-
based work structure.

2.2.4.4. Metadata
The metadata included in any broadcasting file can be either descriptive (format,
copyright…) or active (changing the signal). Loudness normalization can be done by
normalizing the signal during the mixing or also by doing a signal shift before its
broadcasting or before is reproduced using the file metadata.

Of course, the first option is recommended in the EBU recommendation, but it is also
recommended, that the three main loudness measures of a signal loudness have to be
included in a file metadata, those are Program Loudness, Loudness Range and Maximum
True Peak Level. Those three measures are already included in the header of the broadcast
wave file BWF8 (EBU, 2011). For short content programs the Maximum Momentary
Loudness Level and the Maximum Short-term Loudness Level are also recommended to
be stored in the file metadata, as they are helpful dynamics control parameters.

Probably the most used metadata system is the Dolby-Digital system. In the Dolby AC-3
Metadata system, three characteristics of the signal that are of interest for loudness control
are included. The program loudness level, the dynamic range and the down-mix
coefficients. The program loudness is called dialnorm, as the Dolby system is oriented to

8
see EBU Tech Doc 3285; for a detailed description of BWF

25
the anchor signal normalization taking the dialogs as anchor signal, however, this value
refers to the loudness level taking all components into account. The dynamic range
parameter is called dynrng and the down-mix coefficients Centre/Surround Downmix
Level.

The Program Loudness parameter should be set at -23 LUFS when the signal has been
mixed and normalized following the recommendations. If the signal does not fit the
recommendations, the metadata parameter has to be set at the current loudness level of
the signal so the distribution systems can adjust the level live.

There are many situations where the consumer may want to reduce the loudness range of
a program, so many Home Theater systems have a loudness range control option to
control it, but loudness range information is needed in the metadata. So, as the program
loudness level, loudness range can also be adjusted in the distribution system if the
parameters are set in the metadata. In the Dolby AC-3 system different compression
presets are available to fit in different situations.

Also, the down-mix coefficients are important in the loudness level calculation. It is so,
as in the down-mixing the surround and center channels are mixed together with the Left
front and Right front channels in order to have a 2-channel-stereo signal instead of a
surround one, and as explained before, the surround channels are treated differently than
the front channels in the multichannel loudness level calculation. The resulting loudness
level of the down-mixed signal may depend on the down-mix coefficients used, on the
content of the surround channels and on the limiting used to avoid overload in the stereo
channels.

To avoid the overload, good down-mix coefficients have to be used, and also dynamic
processes may be also useful. It has to be taken in mind that those mixes with a lot of
surround content will have a significant variation in the loudness level once down-mixed,
as the +1,5 dB gain factor of the surround channels will not be applied. Those mixes with
less surround presence will not have such a significant loudness level variation.

Metadata can be really helpful for an easier loudness level control, but it also has to be
controlled, as a file with the wrong metadata will produce loudness variations. Therefore,
the EBU recommends to be careful with the metadata, especially in those files coming
from an external source, as it can be set wrongly in porpoise in order to sound louder than
other productions.

2.2.4.5. Alignment signal

An alignment signal is needed in broadcasting in order to set an anchor point for all the
equipment used. This signal is typically a 1 kHz sinewave at -18 dBFS. This level was
specified in the EBU recommendation R68, created in 1992 and revised in 1995 and 2000
(EBU, 2000). This method is not altered with the new EBU loudness recommendations.
It must be said, though, that the signal will be expressed as -18 LUFS, or as -5 LU in the
relative scale, in an EBU compliant meter.

26
 2.2.4.6. Monitoring level
The recommended monitoring level was defined by the EBU in the document EBU Tech
Doc 3276-E ‘Listening conditions for the assessment of sound programme material’ and
its supplement document Supplement 1, for Multichannel Sound. In those documents
some formulas are provided to calculate the recommended listening level, one for a stereo
system and another for a multichannel system.

To calibrate a stereo system, a pink noise test signal at -18 dBFS in digital devices should
be sent to each loudspeaker separately. The gain of the loudspeaker should be adjusted so
that the sound pressure level (SPL) measured using an A-weighted slow response sound
level meter fulfill the next formula in the sweet-spot:

𝐿𝐿𝐼𝑆𝑇𝑟𝑒𝑓 = 85 − 10 log(𝑛)𝑑𝐵(𝐴)

Where 𝑛 is the number of channels of the system. (EBU, 1998)

For a multichannel system the level produced by each loudspeaker of the system in the
sweet-spot should be:

𝐿𝐿𝐼𝑆𝑇𝑟𝑒𝑓 = 96 𝑑𝐵 𝑆𝑃𝐿, referenced to digital Full Scale signal level. (EBU, 2004)

In this case a different test signal should be used. Noise of equal energy per octave,
covering the range from 500 Hz to 2 kHz should be used in this case. The level of this
signal must be the same as in the previous case, -18 dBFS. With this signal sent to each
channel of the system separately, the gain of the loudspeaker must be adjusted such that
the sound pressure level (SPL) measured with a C weighted slow response sound level
meter is:

96 − 18 = 78 𝑑𝐵 𝑆𝑃𝐿

This listening reference levels where defined by the EBU before the loudness regulations
were made. In the R128 recommendation, the EBU recommends not to change the
listening levels or the alignment process. It has to be said, though, that once the
recommendations are applied, the average program loudness level will be approximately
up to 3 LU lower, in comparison with the productions made before the recommendation.
If this level decrease is detected as a problem, a revision of the listening level and the
alignment process will be made in the future (EBU, 2011).

Reproduction Listening level of reference Noise9 rage Level

system
2-channel stereo 𝐿𝐿𝐼𝑆𝑇𝑟𝑒𝑓 = 82 𝑑𝐵𝐴 𝑆𝑃𝐿 20 Hz – 20 kHz -18 dBFSrms
5.1 MCA 𝐿𝐿𝐼𝑆𝑇𝑟𝑒𝑓 = 78 𝑑𝐵𝐶 𝑆𝑃𝐿 500Hz – 2 kHz -18 dBFSrms
Table d. Listening levels and alignment signals summary

9
Noise of equal energy per octave

27
 2.2.4.7. Low Frequency Effects Channel
The EBU recommendation is based in the ITU algorithm, and this algorithm does not take
the Low Frequency Effect (LFE) channel into account due to many uncertainties about
the use of this channel.

This exclusion in the loudness level calculation can cause an abusive use of the LFE
channel. Although it may be included in the loudness level calculation in future revisions
of the recommendation, further practical experience and investigation are needed. (EBU,
2011)

2.2.5. Loudness Parameters for short-form Content

In November 2014, the EBU released the R128s1, a supplement for the recommendation
R128 that concerned additional recommendations for short-form content such as
advertisements, promos or other formats of short duration. This document has been lately
revised and in January of this year (2016) the EBU released a new version of it with some
changes.

In the first version, the supplement recommended, as in the main recommendation, that
the Maximum Permitted True Peak Level of a program should be -1 dBTP and that the
Program Loudness Level should be normalized at -23 LUFS, but this time, with a
deviation of just ±0.5 LU. As complementary measures they recommended a Maximum
Permitted Short-term Loudness Level of -18 LUFS (or +5 LU) or, alternatively, a
Maximum Permitted Momentary Loudness Level of -15 LUFS (or +8 LU). This two
measures were not meant to be applied simultaneously, but alternatively depending on
the needs of the situation.

In the new version, though, there are no longer two alternatives as the Maximum
Permitted Momentary Loudness Level has been left out of the recommendation and now
a Maximum Permitted Short-term Loudness Level of -18 LUFS (or +5 LU) is only
recommended and also, the Program Loudness Level should be normalized at -23 LUFS,
with a deviation of ±0.5 LU. The rest of recommendations and functions stated in the
main recommendation R128 are still applicable.

Programme Loudness -23.0 LUFS ±0.5 LU

Maximum True Peak Level -1 dBTP
Maximum Short-term Loudness -18.0 LUFS (or +5.0 LU)
Loudness Range Not applicable
Table e. Summary of the Loudness Parameters for short-form content (EBU, 2016b, p. 4)

This supplement was created, as it was detected that the measure of Loudness Range was
not effective in this situations, as it is based in a statistical analysis of the Short-term and
for such short programs there are not sufficient data points. Therefore, for such short
contents no maximum or minimum LRA value is specified. (EBU, 2016b)

As explained in the section 3.4.3. Commercial loudness abuse, some broadcasters had
already detected that the recommendation did not deal well enough with commercials and

28
trailers of short duration and they already applied their own additional measures to deal
with this issue.

2.3. Other standards

The loudness standardization is not only an issue in Europe, but all over the glove. Many
different broadcast associations have also created their standards and some countries have
even laws about it.

In the US, the ATSC (Advanced Television Systems Committee) in 2009 created the
A/85, which is a standardization based on the ITU BS.1770. It recommended the
normalization of the loudness level of an anchor signal for regular programs, but with
commercials the recommendation is to normalize the loudness level taking all the signals
into account. In contrast with the EBU R128, the target level is -24 LKFS (instead of -23
LUFS). This anchor signal based method was in 2011 revised and changed for all-source
based measurements for all programs, and the A/85 was then based in the ITU BS.1770-
3, so a gating method was then included as specified in the algorithm from the ITU. (Lund,
2015)

This recommendation was stablished as a rule when the Congress redacted the CALM
(Commercial Advertisement Loudness Mitigation) Act, that directed the FCC (Federal
Communications Commission) to stablish some rules to make all commercials have the
same average loudness level. This rules went into effect the December13, 2012. (Federal
Communications Commission, 2015)

In Japan, the Association of Radio Industries and Businesses (ARIB) created the TR-B32.
The standard for Japanese broadcasters is also based in the ITU BS.1770-2, and as in the
American version, the target level is -24 LKFS and a gating function is included. (Lund,
2015)

In Australia, Free TV released the operational practice OP- 59. This recommendation is
also based on the ITU BS.1770. It also recommends a loudness target level of -24 LUFS
and a maximum true peak level of -2 dBFS. It recommends the anchor signal
methodology, but not with short films such as commercials, where the full mix has to be
measured, also in those circumstances where dialogues are difficult to isolate. (Free TV,
2010)

29
 3. STUDIOS SWR
This thesis will study the implantation and the affectation of the previous regulations and
recommendations in the SWR studios, mainly in the Stuttgart television studios. I will be
performing a 6-month internship in the Stuttgart broadcast studios and during this period
I will be learning how is the practical implementation of those regulations, how to work
with them and what the everyday issues are, in order to provide useful information for
other broadcasters or users.

3.1. Structure
The SWR is the regional public television and radio broadcaster from the south-western
region of Germany, mainly from the federal states of Baden-Württemberg and Rhineland-
Pfalz. Nowadays the SWR is producing programs for several television channels (e.g.
SWR Fernsehen, DasErste, Phoenix…) and six radio stations in broadcasting.

It is a part of the ARD, the consortium of German public broadcasters, which is the
biggest public broadcaster in the world, with 20.616,5 fix employees and a budget of
6.485 € millions last year. (Institut für Medien- und Kommunikationspolitik, 2015)

Regional ARD broadcasters Abbreviation Headquarters city Budget in M. €

Westdeutscher Rundfunk WDR Köln 1.390,4
Südwestrundfunk SWR Stuttgart 1.171
Norddeutscher Rundfunk NDR Hamburg 1.078,4
Bayerischer Rundfunk BR München 1.024,6
Mitteldeutscher Rundfunk MDR Leipzig 681
Hessischer Rundfunk HR Frankfurt 491
Rundfunk Berlin-Brandenburg RBB Berlin/Potsdam 434,6
Saarländischer Rundfunk SR Saarbrücken 117,5
Radio Bremen Radio Bremen Bremen 95,9
Table f. Regional broadcasting stations members of the ARD

30
 Figure xx. ARD structure

With around 3800 employees and 1.171€ millions, the SWR is the second biggest regional
broadcaster in Germany. The SWR has offices in three different cities with ten studios
and 23 regional offices. The offices are in Stuttgart, Baden-Baden and Mainz. Programs
are produced in all three facilities, although the headquarters are in Stuttgart.

3.2. Implementation of the Loudness regulations in the

SWR

As a member of the ARD, the decision to implement the new loudness recommendation
in the SWR, as well as in the other regional broadcasters, came from the ARD. The ARD,
as a European broadcaster, follows the recommendations from the EBU, and as the
loudness regulation was released, they started to study the possibility of its
implementation. After extensive awareness training with experts from different
broadcasters, such as the engineer Askan Siegfried from the NDR (Nord-Deutschland
Rungfunk), who was also participant in the creation of the EBU standard, or from the
engineer Florian Kamerer from the ORF (ÖsterreichischerRundfunk10), who also was a
member of the EBU experts group for surround sound; the ARD, the ZDF11 and the ORF,
together with different private TV stations in Germany and Austria, decided to switch
together to EBU R128 the September 31 of 2012. Several other European and worldwide
countries followed the switch.

Even though the loudness regulation is fully implemented and functional in the SWR
television; in radio and internet is not there yet. Only a few radio station are currently
producing according with the EBU R128. The first ARD radio station to implement it
was SWRinfo in April 2015; Bayern2, a cultural program from BR (Bayerischer
Rundfunk), also implemented the regulations in July 2015, and the rest of radio stations
from ARD will do it too in the future. In internet, material from different standards can
still be found and the recommendations are not a reality yet.

10
Austrian public broadcaster -http://der.orf.at/unternehmen/orf-english100.html
11
ZDF (Zweites Deutsches Fernsehen) – Second German television http://www.zdf.de/

31
Before the implementation of the loudness recommendations, the most of the complaints
that SWR received from the consumers were concerning loudness level changes between
different channels - and between different program parts within one channel. Once the
recommendations were implemented, the amount of consumers’ complaints about this
topic decreased drastically.

3.2.1. Loudness control equipment

The implementation in the SWR TV-facilities in Stuttgart was quite untypical as in 2012,
during the implementation of the regulation, a new production building was being built
and, therefore, thought and equipped to fulfill the recommendation. RTW TM7 and TM9
EBU meters were bought and installed in all the post production workstations (sound
studios, image editing rooms, sound control rooms in live studios, broadcasting room,
technical control, etc.). Also, Nugen loudness plug-ins were bought and installed in the
sound studios and in the image editing rooms. In the other SWR studios, Baden-Baden
and Mainz, the implementation was more gradual, but RTW meters were also installed in
the key positions, mainly everywhere a program is finished and prepared to broadcast.
All those tools are explained below in the section 3.3 My experience with loudness.

At the beginning of the implementation, not all the archive material had the same loudness
level, as it was produced before the regulation, so a very important thing, if not the most
important, is to control that all the programs that are being broadcasted have the right
loudness level. To control that, the SWR installed automatic loudness levelers in the Play
Out Center (POC) in Baden-Baden. Those automatic levelers corrected in real time the
loudness level of the material that had not been normalized yet. A leveler used at that time
was the Inteligain from Evertz. This system was used because the SWR already had
Evertz systems installed, and the only thing needed at that time was an actualization with
the Inteligain system. After some practical experience it was noticed that its performance
was not sufficiently satisfying12.The problem with this and all other real-time systems
was that it was not able to look into the future, which means that it could not preview the
signal and, therefore, predict what gain changes would be needed. Instead, it adjusted the
signal gain during broadcast, taking into account only the present signal. Also the gain
transitions were slow to avoid sudden gain changes.

The system used nowadays is the Minnetonka AudioTools Loudness Control Server. It
levels the loudness of those files that are not normalized before they are broadcasted. The
system is really simple, as only the old files from archive go through this process. This
system takes the video file that has to be normalized, it unpacks the video from the audio,
it normalizes the loudness of the file by adjusting the gain, and it repacks the file together
again. Doing so, the dynamic changes of the mix are respected, and only the general gain
is changed to reach the target level.

In the SWR studios, no loudness metadata is used. The whole production system is
thought to work without loudness metadata, as all the material produced in the studios
should be already being produced as the recommendations state, and therefore no
metadata is needed as no further adjustments will be done. The decision of building a
“metadataless” system was taken at the implementation phase, because it was a much
simpler implementation, and therefore cheaper, than a metadata-dependent system

12
See more in the section 3.4 Daily problems

32
because all the systems used in the production chain should be compatible with the same
metadata and file formats. Also, with no metadata, the mixing methodology must be
changed as the target level has to be reached in the mixing stage, increasing this way, the
quality of the mixes, as there is no longer need to produce over-compressed material in
order to be louder than the competence. The only metadata used in the SWR is the
metadata included in the Dolby-E for 5.1 surround productions, where the down-mix
coefficients must be included.

3.2.2. Loudness in to the workflow

To fully understand the process of loudness control it is important to understand the
production workflow and what are the key spots where the, previous mentioned, loudness
control equipment is used and why. Therefore, we are going to see the production
workflow of a SWR production from its recording until its broadcasting. It must be said,
that this is a standard workflow that may not apply to all products as not all of them have
the same time or resources constrains.

A standard production workflow follows the following steps:

 Recording: the raw material is recorded inside the studios or outside with
recording teams.
 Ingest: the recorded raw material is ingested to the studios server system.
 Video cutting: some parts of the raw material are selected and edited to form
the clip.
 Sound editing: all the sounds and music are mixed and if needed an off-voice is
recorded and added to the mix.
 Image editing: graphics and effects are inserted to the clip.
 Color correction: the color of the clip is corrected.
 Scrutineering: an extensive technical control is done in order to check if the clip
has all the technical specifications needed to continue to the next step, if
not, it must be corrected in the previous steps.
 Archiving: the clip can be stored in the main storage system.
 Preparation for broadcasting: before the broadcast the clip must be loaded
from the storage system to the broadcast server and checked.
 Broadcasting: the clip is send through the play out center.

To start producing material compliant with the EBU recommendation in the SWR, no
significant workflow changes had to be done. The control equipment changed, and
engineers had to pay attention to different characteristics than before, but it is more a
change of habits than a change of workflow. The next list is the workflow again, but we
are going to see where can we find loudness control equipment and why.

 Recording
 Ingest
 Video cutting: RTW TM7 and Nugen Plug-ins are used, as some clips are
finished in this stage because they are not significantly complex and there
are big time constrains (news), so there is no time to do an extra sound
editing. The RTW TM7 is used to control that the final product reach the
targets of the recommendation. The Nugen Plug-in is sometimes used to
normalize the loudness level of old material from the archive before to use

33
 it and/or mix it with new material. If the material is not finished in this
stage, the loudness level is just approximated to the target, as it is not the
final product yet.
 Sound editing: RTW TM9 and Nugen Plug-in are also used. All the sounds,
music and voices are mixed and it is normally the final stage for sound
edition. There are clips that are only part of another program, e.g. clips that
are going to be played live in a TV show. Those clips are just approximated
to the target level, but they have a deviation usually never bigger than ±1
LU, as the engineer in the live control will adjust the final volume of the
whole program to reach the target. If the material that is being mixed is an
entire program ready to be sent to the POC, the target is tried to be reached
while mixing, but if there is a small deviation, the Nugen Plug-in is used
to adjust all the parameters.
 Image editing: RTW TM7 is also used here, as some of the graphics may contain
a sound effect too and also because the editing rooms are also used for
video cutting.
 Color correction
 Scrutineering: A RTW TM9 is used to control that the production has the
technical specifications needed. If the specifications are not fulfilled they
must be corrected. Depending on the time constrains, the product can be
edited again, but if there is no time, an automatic loudness corrector will
adjust the signal level before the broadcasting.
 Archiving
 Preparation for broadcasting: All the material produced before September 2012
has to be corrected before it is broadcasted. It can be done manually in the
sound studios, but if there is no time, an automatic loudness corrector is
used. The current loudness leveler used is the Minnetonka AudioTools
Loudness Control Server, which analyses the file and adjusts the gain of
the file to reach the target level.
 Broadcasting

As seen, a special case that has to be taken into account, is the procedure to follow with
the old material, which is not loudness normalized. When a file from the archive is used,
it is immediately normalized, in order to be able to mix it with new material without
loudness level differences. If the file that has to be used is a whole program, the
Minetonka server in the Play Out Centre analyses de file and normalizes the loudness
level before its broadcasting.

3.3. My experience with loudness

During my praxis semester I was performing different functions in the broadcaster, which
gave me the opportunity to have a general vision of the whole production chain, and I
could see how and where the loudness level recommendations are applied.

3.3.1. Loudness in live broadcasting

My first experience in the SWR studios was in the live productions, such as magazines,
political debates, news, etc. I was first learning the methodology in the studio and also in
the control room.

34
The in-studio work consists basically in establishing communications between the control
room and the studio, activating all the necessary microphones and loudspeakers, rooting
all the necessary signals from and to the control room, and checking that all work
perfectly all the time. We are not going to focus in this part as it is irrelevant as loudness
regulations are concerned.

In the control room, the sound engineer has to take all those signals from the studio and
mix them live, and of course, the final result should have the correct Loudness Level,
Loudness Range and Maximum Peak Level. To achieve it, the sound engineer has
different tools to use.

The sound mixer used in the live control is the Lawo mc2 66. It is a professional mixing
console from the German brand, Lawo. The model used in SWR Stuttgart studios has
32+8 faders that can be multiplied as there can be up to 6 banks with two different layers
each. It is based in a routing system between the console and the processing unit, with
redundant inter-connection paths. It can be accessed from an external computer in order
to assure full control at all time. It is compatible with mono, stereo and surround systems
up to 7.1, also with Dolby E, and it has an integrated loudness metering system compliant
with ITU-R BS.1770. There are many more features to talk about, but we are going to
focus in the loudness metering systems that are integrated in the console.13

The mc2 66 it selves has an integrated metering system compliant with ITU-R BS.1770,
which is largely compatible with EBU R128 and ATSC A/85, but as a European
broadcaster, in SWR the EBU standard is used. That means that the metering offers
momentary, short-term and integrated loudness measures, and also in two different scales,
+9 and +18.

What I saw in my practical experience is that the display of the loudness measure can be
differently configured. The configuration in SWR Stuttgart studios is the next: in the
central console panel the integrated loudness value can be shown in a big central number,
so the sound engineer is fully aware of the level of the program at all time, the sound
engineer can chose which parameter to choose (Loudness Level, time code..), and the
loudness level can also be shown in LUFS or as a relative value in LU. The dynamics
processes, such as compression, that are applied to the signal are also shown in the main
display. Also, the momentary, short-term and integrated loudness measures can be shown
in display, but normally the external computer that is also connected to the console is
normally showing those levels in an external display. The scale used is the +9 LU scale
as it has enough dynamic range for broadcasting. All those parameters can be changed
and al the displays can be fully adapted to the engineer’s preferences.

13
More information can be found in: https://www.lawo.com/products/audio-production-
consoles/mc266.html

35
 Figure xxi. Main display of a Lawo mc2 66 from SWR Stuttgart studios with loudness meter in LU

Of course every signal, which are distributed in the console faders, has its own meter
mainly to see graphically what signals are present at that moment, the meter chosen by
default in the SWR Studios is a peak meter with a peak hold of 3 seconds to control that
the signals do not have too high peaks. The meter for every fader can be changed to the
next different options: the previously commented peak meter, a fast peak meter with a 1
ms integration time, a momentary loudness meter, or a VU meter.

Figure xxii. Parameters in the Lawo console meters

36
The signal that goes through the EBU R128 compliant loudness meter is only the final
mix of all the signals, as this meter is meant to measure the loudness of a whole program.
In the next figure we can see the screen of the external computer linked to the mixing
console, where EBU compliant meters are shown. The one with the three scales is
metering the final sound of the program which is being broadcasted. All the others have
just the M scale for both channels of the stereo signal. This structure can be fully adapted
to show different parameters for each signal.

Figure xxiii. External computer for Lawo mixing console monitor

The mc2 66 has yet another EBU meter integrated,

but this one is independent of the others as in fact it
is a RTW TM7. The RTW TM7 is a touch monitor
which normally is a separated module, but in this
case, thanks to a business deal that was announced
in 2010 in the NAB (National Association of
Broadcasters) show in Las Vegas (RTW GmbH &
Co. , 2010), it is fully integrated in the Lawo mc3 66
just next to the main display.

The touch monitor is fully adaptable to every sound

engineer. There are different presets with different
configurations in order to reach the target easier.
Some sound engineers like to see all three meters in
the display (M, S and I), some others prefer to see
just the M and S measures as the integrated one can
also be seen in the main display. All this presets and
configuration data are stored together with the
project session of the Lawo mixing console for
Figure xxiv. RTW TM7 from Lawo mixing
console
every sound engineer in the house.

37
Even though the two metering systems (Lawo and RTW) are independent, they normally
do not differ for more than 0.1 LU from each other, and it is mainly because they are not
reset at the exact same moment.

With this redundant adaptable metering system, sound engineers do not have problems to
reach the -23 LUFS target level. As usually before the live program starts all microphones
are tested, and the levels are pre-adjusted depending on the presenter. Also, all the music
pieces and external videos are checked and small notations are made depending on how
loud every signal is, because attenuation or an increment of the signal level might be
needed when played live in order to reach the target. However, all reportages produced
in the SWR should all have approximately -23 LUFS as integrated loudness level, so no
further big adjustments should be needed when playing them live.

Before the program starts, there is another thing to check. The SWR has studios in three
different cities, and before a program starts in one of those studios, the connection
between the studios and the Play Out Center has to be checked. To do so, the SWR does
not follow exactly the EBU R68 recommendation. Instead of 1 kHz at -18 dBFS, the SWR
uses an escalating signal of 1 kHz starting at -30 dBFS, after two seconds it increases the
level until -18 dBFS, and after two seconds more it increases until -9 dBFS; after two
seconds more it starts again. This way, the EBU recommendation is integrated, and an
extra parameter is checked. By increasing the level of the signal the engineers can also
detect if there is an unwanted dynamic process in the connection path between the two
stations.

3.3.2. Loudness in sound production and post-production

Once I learned how the studio work was done, my next step in my praxis semester was in
sound production and post production. There, small clips for different programs are
usually mixed, but also bigger productions are done. The material comes ready to mix
from the video cutters, where the producer already selected the ambient sounds and the
music.

The system in postproduction is from Avid, and it consists on a Media Composer

computer connected via Interplay to the ProTools computer. The two systems need to be
connected as they have to work together in order to provide video and sound at the same
time. They do not have a master-slave relation, instead, they just run the files together.

The ProTools used is the ProTools 10 HD, which is controlled by a mix console from
Digidesign. This mix console is, in fact, not a mixer but a remote control from the
ProTools system. There are many sound engineers in the house and each of them has an
individual ProTools session with different fader distribution and screen memories.

Once all the systems are on and working properly, the mixing can be started. In order to
achieve an appropriate sound level, ProTools has its usual peak meters to control that no
signal is clipping, but the most important metering instrument is the TM9 from RTW.
This is a touch monitor really similar to the previously explained, TM 7 incorporated in
the Lawo mixing console, but this time it is a separate module. It is a 9 inch touch monitor,
with a flexible configuration graphical user interface, on which the engineer can chose
what parameters to display. The configuration used in the SWR consists on an 8-channel
true-peak meter in dBTP, the three loudness scales (M, S and I) in LU, and a spectrogram

38
as main indicators, a small phase and correlation display in the right top corner, and in
the left bottom one, we can see the LRA, maximum TP and S values displayed.

Figure xxv. RTW TM 9 touch monitor from a sound editing room in the SWR Stuttgart studios

Before my praxis semester I did not have much mixing experience and no experience
with loudness meters at all. Even though, it was not difficult for me to get a good mix that
reached the loudness level target after a couple of indications from the engineers.

The first indication given was: “You have to mix with your ears”. I had never done it
before, but it was quite intuitive. Everything had to sound natural, and there was freedom
for the loud parts to be loud and the soft parts to be soft. I just had to make sure all the
elements present in the scene were coherent with each other, and that the music added
transmitted what the producer or author wanted to transmit.

Secondly, a good monitoring level is needed to let the loud parts be loud and the soft ones,
soft. In the SWR studios there is no predetermined monitoring level. Instead, every sound
engineer has its own monitor gain level depending on their preferences, and audibility
capacities. I noticed that if the gain monitor is too low, the loudness meter indicates that
the mix is too loud and vice versa. So, as we are mixing with our ears, the monitoring
level is very critical. The monitoring system used in the SWR is a Genelec 5.1 system
that consists in five 1238CF (with DSP inside) SAM™ Studio Monitor and a 7270
SAM™ Studio Subwoofer. This system is used in the post-production and live studios
and it was calibrated with the Genelec auto-calibration technology AutoCal™ and
Genelec Loudspeaker Manager (GLM™) control network technologies. Every monitor
in this system is connected with each other creating a network and the system
automatically align every monitor on this network in terms of level, timing, and
equalization of room response anomalies. (Genelec Oy, 2015)

Finally, the third indication was that speech intelligibility is very important. I had to make
sure that the voices in the films were understandable for everybody, and in any audition
environment. This is probably the most important one, as without intelligibility the films
lose their purpose: to communicate.

To make sure that all the dialogs were understandable, I just placed the voice loudness
level in my center of reference, which was -23 LUFS. That means, that I just made sure

39
that all the voices had an average level of -23 LUFS, and let the other elements free around
the speech level. This technique is discussed in previous parts of this work14, and also in
the EBU 3343 document, where the reference signal is referred as anchor signal. To get
the anchor signal level right, I always adjusted its level depending on de value of the
Short-Term loudness meter. I normally did not look at the Momentary meter, as it did not
give me any relevant information, as the voices should have momentary level fluctuations
to be expressive and sound natural. In order to assure speech intelligibility, filters were
also applied to the voices, to filter all the frequencies that may disturb. This methodology
works normally, but sometimes it presents problems in some situations15.

By doing all those things, the target of -23 LUFS was normally reached or nearly reached,
with a deviation of normally no more than ±1 LUFS. I also noticed, that when mixing
short clips, the deviation is bigger than when mixing long films (half an hour
approximately), where the target was usually perfectly reached. When the target is not
perfectly reached there are several options. If the piece that we are mixing will be played
back in a live production, e.g. a magazine show, no further adjustments are needed, as the
clip is going to be sent through the mixing console of the live studios, and the sound
engineer will adapt the level of all the signals to reach the target of the whole program.
But, if the piece that we are mixing is a whole program ready to broadcast, then we have
to adjust the loudness level to reach the target perfectly. To do that, the SWR has a
ProTools plug-in that analyses and adjusts the average loudness level of our mix.

The plug-ins used are the NUGEN Audio LM-Correct. It is a loudness analyzer and
corrector, capable of working up to 100 times faster than real time, compatible with mono,
stereo and surround up to 5.1 files. It is compliant with EBU R128, ITU BS.1770 and
CALM Act. Momentary and Short term maximum levels can be set. A LRA target is also
optional (NUGEN Audio, 2016).

In the SWR studios this plug-in is set to be compliant with the EBU R128, that means
that the loudness target level is set at -23 LUFS and that the True Peak maximum level is
at -1 dBTP. The application analyses the mix faster than real time, and indicates if the file
should be adjusted or not. Then by just pressing render, it automatically adjusts the values
in order to meet the targets of the regulation. Extra parameters, such LRA or Maximum
Momentary or Short-term Loudness Level, can be also adjusted, but in the SWR studios,
those options are normally inactive. Normally four stereo tracks are saved with each clip,
but only the main one is normalized as it is the only one that will be broadcasted.

14
See 2.2.4.2 What to measure
15
See 3.4 Daily problems

40
 Figure xxvi. Nugen Audio LM-Correct plug-in interface (NUGEN Audio, 2016)

The four stereo sound tracks are the next: the main one, is the ST (Sende Ton) which is
the final mixed sound of all the elements ready to broadcast. The second one is the IT
(Internationaler Ton), which is all the ambience sounds and voices present in the video
and all the background music. The third one is the IT ohne Musik, which consists on the
same elements of the IT sound but without the music. And finally, the fourth track is the
off voice, where only the off voice recorded in the studio can be found. Every small clip
is saved like this in order to be able to reuse the material in any situation, as the video
with its ambience sound is saved, but without the music or off voices.

When a whole program is produced, the four stereo tracks of the file are organized
differently: the first one has the ST sound; the second one the IT sound; the third one,
however, has the audio-description track; and the last one has the Dolby-E information
for surround systems. Only in the clean feed16 version of the program the organization of
the sound tracks is as explained in the previous case (ST/IT/IT without music/Off-voice).

3.4. Daily problems, abuses and shortcomings

All changes need a transition time, and during this time problems are detected and
corrected until the optimal situation is found. The implementation of this regulations
suppose a change of habits and of course many subjects were discussed deeply as some
shortcomings were detected in the first approach.

All the next situations, and examples are real situations experienced by myself or by
sound engineers with which I had the opportunity to discuss and share opinions and
experiences during my praxis.

16
A clean feed version of a program consists in the same program, but without all the graphics or color
corrections made or inserted in the recorded video.

41
 3.4.1. Automatic loudness level systems

One of the problems detected concerned the automatic loudness levelers of the Play Out
Center in Baden-Baden. The system was the, before mentioned, Inteligain from Evertz.
This system was not able to see the loudness changes before they happened, and it only
changed the signal gain when it detected that the signal was too loud at that moment. This
caused several problems in the next situations:

 Loud-soft transition

When the real time leveler detected that a program was too loud (e.g. -18 LUFS), it starts
reducing the gain until the target level is reached (e.g. -5 LU to reach -23 LUFS). Then a
file with the correct loudness level comes and, as it cannot predict the level of the next
file, the gain is still at -5 LU. Therefore, the next file will start at -28 LUFS. Because of
the slow transitions in order to avoid sudden loudness differences, it takes a while until
the level reaches the target again. This causes that, during this transition time, the
consumers have already used the remote control to turn up the volume as the beginning
of the program is too soft. When the gain is adjusted again, the consumers have to use the
remote again to turn down the volume as now it is too loud. As we can see, this causes
the reverse effect of the aim of the regulations, the consumers have to adjust the
reproduction level two times as the gain adaption is too slow.

Figure xxvii. Graphic representation of a loud-soft file transition

 Soft-loud transition

The same happens with the opposite situation. When a file is detected to be too soft, the
leveler increases the gain to reach the target. If a louder file comes next, the gain is still
increased, and therefore, the file is even louder at the beginning. By the time the leveler
detects the loudness level and decreases it, the customers at home have already jumped
from the sofa and turned the volume down. Again, when the leveler has reached the target
level, the customers have to use the remote again to turn up the volume as now it is too
soft.

42
 Figure xxviii. Graphic representation of a soft-loud file transition

Of course this situations caused some troubles to the customers, but also to the sound
engineers who saw their mixes changed in a bad way. Those situations happened in real
life, below we can see a graphic obtain by measuring the signal of a broadcaster.

Figure xxix. Measurements from actual loudness transitions during broadcasting

This only happened at the beginning of the implementation, as the decision to change the
leveler was taken. The current method used to normalize the incorrect material is quite
different. The system used is a Minnetonka system, and as said before, only the files that
have not been normalized go through the normalizing process. This method is much better
than the old one, as respects the dynamics of a mix and only adapts the gain of the whole
signal. The only problem of this method, is that it has to be done before the file has to be
broadcasted and it is time consuming. Because of the complexity of the unpacking and

43
packing process the system works practically in real time. That means that if a program
of 90 minutes has to be normalized it will take almost 90 minutes to be done. This may
suppose a problem, as this calculation time has to be taken into account, because the file
has to be ready to broadcast at the right time.

Another problem related with the calculation time, was faced at the beginning of the
implementation. At that time, a lot of not normalized material had to be broadcasted, as
almost all the archive material was produced before the regulation. This situation caused
that the leveler had to be constantly working and sometimes the time was tight.
Nowadays, most of the programs broadcasted are already produced according to the
recommendation and only one or two files per day have to be normalized.

3.4.2. Tricky situations

Despite all the techniques and technologies explained in this study, there are some
situations in which reaching the target level may be a little bit tricky, or if reached, the
result may not be very satisfying. In some situations, also, sudden loudness level changes
may also occur despite following the recommendations. Here, I present some of these
situations:

In films there are typically more dynamics than in broadcasting, and therefore, there are
very loud parts and very soft ones. In broadcasting, though, there are not that much soft
parts, as the listening environment does not permit it, but also because the material
broadcasted is mainly talk-shows, news, magazines, etc. In this type of programs the voice
has the main role, and some music and effects may also be added to it.

As explained before, it is intended for the voice to have a level of -23.0 LUFS in order to
assure its speech intelligibility and because it is the main element present in almost every
production. This cannot be done in all situations for the target to be reached, and may
cause some trouble. The next situation may be an example of it:

 Voice loudness level differences

Imagine a talk show where there is a moderator that presents the show and also a guest
that talk about some daily topics. There are some different sections in the show and
between each part an energetic music piece is played. This music should be loud in order
to have some presence, to keep the attention of the viewer and to fill those transition
moments in the show. In order to be coherent, the music should be definitely louder than
the voices, but that may present some trouble. If the voices are leveled at -23 LUFS, and
the music should be louder than the voices the target level cannot be reached.

44
 Figure xxx. Program scheme with -23LUFS voice level

Therefore the levels have to be adjusted to reach the level but also to maintain the
coherence between the voice and the music. Now the voice is softer than -23 LUFS to
leave some space to the music to be louder, and the target is perfectly reached.

Figure xxxi. Program scheme with -23LUFS program level

Even though the target is reached, this may cause some troubles because, as said before,
voice is common in almost all the productions. That means that the viewers that may
change the channel while watching this program will perceive a loudness level change
between both channels as the voice of the first program was softer than the second one.
This is the reason, that some engineers would prefer the dialog level loudness
normalization instead of the EBU R128. Also some studies has shown that consumers
prefer a constant dialog loudness level rather than a constant program loudness level.
(Carroll, Jones, & Williams, 2007)

45
 Figure xxxii. Voice level differences between programs

In another particular situation a similar problem occurs. When mixing a sports program
(normally football), it is typical to have the best moments of the game, with some
comments of an off voice added, and after, some comments of the trainer or players during
the press conference with no more ambience added. As said before, it is wanted that the
voices have a level of -23.0 LUFS in order to assure intelligibility. Here, the main voice
is mixed together with a football stadium ambience sound, so it will be mixed softer, as
the sum of both (ambience and off voice) should be near -23.0 LUFS as it is the main part
of the program, timely speaking. The comments of the trainer during the press conference,
however, have no added ambience and can perfectly be mixed at -23.0 LUFS. The
problem here is that the two voices will have different loudness levels, and this difference
will be perceived as a loudness level jump by the consumer, even though the two parts of
the program have matched the target. It is also incoherent as the main voice should be the
off voice in this case.

 Sudden loudness level changes

A situation where I realized that the loudness level suddenly jumped was while watching
an action movie in a private German broadcaster. Action movies have normally more
loud parts than other kinds of movies, as there are lots of shots, explosions, car chases
and loud music. As the average loudness level of those films must be -23.0 LUFS when
broadcasted, the soft parts of actions movies (e.g. dialogue) are very soft in order to reach
the target. What happened while watching the film, is that there was a commercial brake
during one of those soft parts. Of course, the commercial was played at the right loudness
level, but it sounded very loud in comparison with the soft part of the movie.

46
 Figure xxxiii. Loudness level difference between a film and a commercial

One more example of a sudden loudness level change can occur in the transition between
two programs. Imagine a program is being broadcasted live, and the sound engineer
realizes almost at the end of the program that he is mixing too soft and that he still has
some “space” to mix louder. Then, in order to create impact and to enhance the final
music of the show, he mixes the end of the program louder. When the next program starts,
it will be much softer than the previous one, as the ending and beginning of the programs
have very distant levels. A further recommendation might be that the beginnings and
endings of the programs should be not too distant (±1.0 LU for example) from the target
level in order to avoid these kind of loudness level changes.

3.4.3. Commercial loudness abuse

It is well known that the main problem, before the loudness regulations, were loud
commercials. Now, with the new recommendations, this fact has changed significantly,
but there are still some cases that commercials take advantage of the regulation.

A particular case of this situation are the drugs commercials. In these commercials there
is a safety message that states that the consumer should read the drug precautions and
should talk to the pharmacist before taking the drug. This message is typically at the end
of the commercial and it has an approximate duration of 4 seconds. It may not seem much,
but in an advertisement it is a significant amount of time. The strategy of this commercials
is to make the commercial content louder than the target level and making the final safety
message much softer so that the average loudness level is -23LUFS. By doing this the
drug commercials can be louder than others producing again loudness level differences
between programs.

Because of some of this situations some broadcasters are applying some extra regulations
to their content, as the EBU R128 is just a recommendation. Some have add a maximum
Momentary and Short-term level, some others have a maximum Loudness Range and/or
True Peak level, depending on their preferences and needs.

47
 4. CONCLUSION
In the 2000s the loudness war was causing struggles, also in broadcasting, with
uncomfortable loudness level differences and very loud commercials. To put an end to it,
loudness level recommendations were created and they have been one of the most
significant changes in the history of broadcasting sound. Thanks to a lot of effort from
regulatory institutions, broadcasters, manufacturers and professionals, the
recommendations from the EBU R128 are nowadays a reality in many European
television and radio stations. Since its creation a big debate started in all broadcast
companies to figure out if it was, or not, a good solution for the loudness level differences
that were causing many complaints at that time.

In the SWR, and in Germany in general, it was decided to take a step and make a change
in 2012, but to do it new technology and knowledge was needed. In the SWR a relatively
simple solution was found for its implementation, proving that it is possible to follow the
recommendations without making big significant changes in the workflow or in the
equipment used in production. To make it possible, the professionals of the house had to
be trained as the mixing methodology had to change. Also, some equipment had to be
purchased, such as EBU compliant meters, plug-ins and automatic loudness levelers. As
time passed, experience was acquired and some problems or opportunities to improve
were detected, so some adjustments to the original plan were made and some equipment
was changed.

Now, after four years of its implementation in the SWR studios, the recommendations are
fully implemented and integrated in the daily work of the engineers. After many hours of
discussion with the professionals we concluded that, although no rule can be perfect for
all situations, the EBU R128 is a significant step in the right direction. There are some
situations where the recommendation does not adjust well enough, but it was also
mentioned that those situations also happened before the recommendation was
implemented. Because of this situations, there are some professionals who think that a
dialog level normalization, together with loudness range recommendations would adjust
better to the needs of the broadcasting industry, and that after all this time, revising the
topic would be a good idea.

It has been detected that these recommendations are not being applied in all European
broadcasters and that there still exist loudness level differences between programs and
channels. Therefore, in my opinion, a similar model as the one implemented in the USA
would help end up with this situation, as it is mandatory by law (CALM Act) to maintain
a specified integrated loudness level.

Radio senders are a good example of broadcasters that have not implemented the
regulations yet. This thesis focuses only in the implementation of the loudness level
recommendations in television, but a very interesting extension of the topic would be its
implementation in the radio. This has not been covered here because of time constrains,
as the dynamic processes in the radio are relatively complex. Therefore, I strongly
recommend future researchers to write about this topic. Also, the SWR is a television
channel with no commercials, and it would have been interesting to see how commercials
are treated before their broadcasting. It would be also interesting to focus in surround
sound productions, as more parameters have to be taken into account while mixing in
order to reach the target level. This topic is not extensively covered in this thesis, as I had
not the opportunity to work in any surround sound production during my praxis semester.

48
With this thesis I increased my knowledge about the topic. I also gave out, put together
and summarized relevant information for other broadcasters to implement the
recommendations. Furthermore, I pretended to increase the awareness of the consumers
too, as almost everyone has a TV at home and can detect loudness level differences. After
reading this thesis, everyone should be able to understand what is happening when a
loudness level change is detected, and I hope that this will help consumers choose and
appreciate the content they want to see and, of course, to hear.

49
 5. TERMINOLOGY
[1] Dynamic Range: difference of a signals level between its softest and loudest.

[2] Crest Factor: It was known as the difference between the highest peak of a signal and
its average level, measured in dB. With the apparition of the program loudness, it can also
be defined as the difference between the signals highest sample peak and the average
loudness level.

[3] Sample Peak Level: the highest absolute numeric value of the samples of a file.

[4] True Peak Level: the level of a peak of a signal detected with a True Peak meter by
upsampling the signal and estimating the inter-sample values, being able to detect
intersample peaks, which are often higher than sample peaks

[5] Intersample Peak: Additional peaks that are not represented by the samples of a file
that can occur between samples when filtering, changing the sample rate or playing the
signal through a DAC. They are normally higher than sample peaks and they are common
in highly processed material with lots of compression, limiting, etc. They can be detected
with a True Peak meter by upsampling the signal. That is why they can also be known as
True Peaks.

[6] Headroom: The space (measured in dB) between the highest peak of a signal and the
level where clipping and distortion starts to occur due to overloading.

[7] Leq(RLB): measure of the average energy of an audio signal in a period of time using
the RLB (revised low-frequency B-curve) weighting curve.

[8] K-filter: Filter used in the ITU BS.1770, it is implemented in two steps. First, a pre-
filter that consists in a shelving filter that boosts the high frequencies to compensate the
acoustic effect of our head, and then a high-pass filter, known as RLB (Revised Low-
frequency B-curve), that approximates the human loudness perception depending on the
frequency band.

[9] PLOUD Group: The loudness project group with over 240 participants including
creative and technical experts from the European Broadcasting Union (EBU), creators of
the EBU R128 Loudness Recommendation and all the complementary documentation.

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