Posts mit dem Label mixdown werden angezeigt. Alle Posts anzeigen
Posts mit dem Label mixdown werden angezeigt. Alle Posts anzeigen
Sonntag, 19. Juli 2020
LIve Online Stream Mixdown Session
It wasn't just through Covid 19 that I came up with the idea of offering my customers the opportunity to experience the mixdown of their title live over the Internet. Via YouTube he gets a look at my screen, in the studio (my person) and of course there is the live sound in a very good quality. Via chat there is the possibility to intervene live in the event and to help shape the sound. The customer hears his title in his familiar environment and can thus judge how the title sounds to his taste. The prerequisite for this is an Internet-capable computer with an audio card, connected speakers (or headphones) and a YouTube registration for the chat. Here's how it works: An appointment is made in advance. The customer will receive a YouTube URL by email on this date. If he enters / copies this URL into his internet browser, he will hear his title directly and be able to see the mixdown. He can contact me via chat via the Youtube platform. In most cases I will not start the mixdown from scratch, but there will be a pre-mix that can be played. In the course of the title, he can now express change requests that I implement directly. The session ends when the customer has found his sound. Stefan Noltemeyer www.mastering-online.com
Donnerstag, 7. November 2019
One Master in two different levels
Now our clients will receive their masters in two different maximized versions.
Version 1 is mastered without a noticeable limiter. This version is used for the online platforms. It has a much higher dynamic than version 2. The second variant is maximized as usual, as it is well-defensible and loud.
to 1: Most online music platforms play their songs at a level of -18 to -14 lufs. Broadcasting establishes a level of -23 lufs (R-128 norm). Loudness Units Full Scale is a frequency weighted unit for loudness. The loudness is determined over the entire title. It makes no sense to create a master that is initially maximized to reduce it by at least 6 dB. So we have decided to create masters which are practically without noticeable use of a limiter. It does not depend one litte dB, because the level is also determined by the arrangement of the song. If a title contains quiet passages, then averaged the loud sections may be a bit louder than if the level were relatively consistent. It is important that this master does not influence the sound by limiting or maximizing. We welcome this development.
to 2: The level on a cd is another topic. Depending on the genre, it still gets really loud here. With a cd master we proceed as usual with a level as loud as possible, without "overdoing it". Of course, it may also make sense to use a limiter as a sound-influencing element to deliberately create a sound of "compression". Stefan Noltemeyer www.mastering-online.com
Mittwoch, 16. August 2017
10 Tips for using an audio compressor in the mixdown.
1.
The best audio compressor is the one whose presence is not heard
at all. (Apart from exceptions such as side-chain compression in
EDM).
2.
A compressor does not increses the signal, but it reduces peak
levels. Only the Make Up Gain will raise the level.
3.
This does not make the loud sections of the music louder but the
quiet.
4.
The compressor reduces the original dynamics.
5.
Problematic are the transients (fast attacks), for
example of an acoustic guitar. The transients compressed at first, because they have
a relatively high level. This reduces the level, but also the
original sound, because transients are very important for the sound.
6.
Voices can be easily reduced because they have very few transients. 12 dB and more (recommended ratio 4: 1) are pssible.
8. With the side-chain input, the
work of the compressor is not controlled by the input signal, but
by the signal that is present at the side-chain input.
9. For example, a synthesizer in a mix
can be automatically quieter when the voice is applied to the
side-chain input. This creates space for the voice and
makes a mix lively.
10.
In the parallel compression, the compressed signal and the
uncompressed signal are interconnected phase-stiff. This way you can increase the low level signal and keep the original transients. (Dry control)
Stefan Noltemeyer
www.mastering-online.com
Stefan Noltemeyer
www.mastering-online.com
Freitag, 28. Juli 2017
technical anaysis of a produced song
Before I start mastering a song, I listen and analyze its acoustic-technical problems. I separate the overall sound into its components.
First, I check out the low frequencies.
Are the basses generated by the bass,the bassdrum or both?
Are there any other instruments which sounds below 100 Hz? -
If the Lo End o.k or should I use a high-pass filter above 30 Hz. (This shows me also my spectrum analyzer)
Are there any problems between the bass and the bassdrum?
Next, I check out the mid frequencies between 200 Hz and 3000 Hz (where the "music plays").
Is there any instrument or a voice that is too loud?
Is there a single frequency that is too loud?
Is there any instrument or a voice that should be louder?
Is the kick of the bassdrum (the attack) and snaredrum powerful enough?
What about the presences (between 3000 Hz and 8000 Hz)
Which instruments are still involved in this frequency range?
Is the voice sound present enough?
Are there problems with "S" in the voice?
Are there instruments and overtones that are to strong here?
Is the hihat too loud (classic error in the mixdown)?
Is Snaredrum still there?
Next check out the high-end above 8000 Hz
Which instruments play "top"?
(Again) Are the hihat, ride and crash o.k.?
What about the total sound
Does anything boom in the bass?
Does the title sound too sharp or too dull?
What happens in the side channel, are there phase cancellations (stereo information)?
What is the total level, has the sum ever been compressed?
Do we need more loudness
if I´m unsure I will listen to the song in comparsion to other titels same genre.
Stefan Noltemeyer
Donnerstag, 10. Dezember 2015
Linear and non linear audio distortion
Essay
from the book "Mastering" (translated by google)
Stefan
Noltemeyer: www.mastering-online.com
Email:
info@mastering-online.com
Distortions
All
components of a sound studio can be divided into four groups:
1.
transducers (microphones, speakers)
2.
amplifiers (preamplifiers, power amplifiers, impedance converter,
equalizers, compressors, limiters, effects units)
3.
Memory (analog and digital tape machines, computer hard discs)
4.
Cable
Excepting
the digital storage, all these components produce distortion in
varying levels.
Distortions
inevitably arise in any type of audi transmission.
Distortion
in the audio technology means , the modification or manipulation of
an original signal. Distortion may intentionally or unintentionally
caused.
One
distinguishes between linear and non-linear distortions.
Linear
distortions
In
linear distortions can turn different delay distortion and amplitude
distortion. In contrast to non-linear distortions incurred in linear
distortions no additional frequencies or harmonics.
Runtime
distortions in each transmission chain. A signal always comes later
to the receiver, as it has been sent. If a full-range equally offset
in time, the problem is mostly irrelevant. Exceptions are the latency
problems in working with computers. The processing time required by a
DSP (Digital Signal Processor), audio data can be processed depending
on the processing power and quality of the audio card to several
hundred milliseconds, respectively. When a musician with his
instrument or recording his voice "by the computer" listen,
he will feel the delay as disturbing.
In
the analog audio technology problems arise when several signals with
different delays are switched together.
Delay
distortion in which the delay is shorter than the period of a single
vibration, called phase distortion or phase shift.
If
different frequency bands vary greatly delayed in a transmission
chain, may lead to audible effects.
I
distinguish two different groups of perpetrators.
Firstly
produce individual components in the electronics delay distortions,
on the other hand caused delay distortions under certain conditions
in acoustics.
In
an amplifier stage, there are different times for high and low
frequencies by specifications of individual components. A well-known
example is the so-called RC member. It consists of a combination of a
capacitor and an electrical resistance. Such a RC member, for
example, the main component of an equalizer. It shifts the phase
angle from high to low tones within an audio signal. Audible this is
only in extreme cases. Relevant it is when adding up many small
runtime shifts. Details are described in the chapter "Equalizer".
On
closer examination of the acoustic aspects of audio engineering, one
often encounters phenomena that cause delay distortions.
Phase
cancellations make disturbing especially with low- frequency signals.
In
a graphic example, we have two sine waves with a frequency of 50 Hz
and amplitude equal to 10 milliseconds large relative to one another.
This corresponds exactly to a 180 degree phase shift. The vibration
is placed in the left channel of the stereo image, the other right.
If you switch now the signal to mono, the two vibrations are added to
a sum. At the crest of a sine wave is a wave of the other sine wave
added. In this case, delete them both completely made Dis sum is
zero. Silence !
Since
there is practically no music in the pure sine waves, and as a phase
shift of exactly 180 degrees will never happen, this is a theoretical
example. At a wavelength of several meters for low frequencies (6,80m
at 50 Hz) can very well keep in mind that the phenomenon of phase
cancellation blatantly changes if, for example, one meter further
away from the sound source.
When
you record a sound source with two or more microphones,
it
comes to phase cancellation, since the sound requires different
maturities to the different microphones. It caused interference when
you switch the signals from the microphones together. Clear up the
sound waves, is known as destructive interference, add up the sound
waves it is called constructive interference. In both cases there is
a distortion of the original sound. In acoustics it is called comb
filter effects.
The
frequency spectrum of an instrument e.g. an acoustic guitar or a
wing, which is picked up by two microphones, by such a comb filter
effect is partly strongly distorted. The decisive factor is each
other the volume balance of both microphones.
Are
they identical, the effect is strongest.
Is
a microphone, for example, one meter further away from a sound
source, as a
otherwise,
the result is a delay difference of 0.029 sec.
Almost
classical problem is during the recording of a piano or a grand
piano. The problem arises as follows: It is common that a microphone
for the bass and one is used for the treble. Now since both
microphones not only the area for which they are intended, to record,
but the full range of the wing, there is phase shifts due to
different maturities of the two signals. Since the sound is very
slow, small run-time differences become a problem already. To
illustrate this, let us imagine a single bass string of a wing. If
she was injured, the resulting sound from two microphones is
recorded. Now these two microphones have a different distance from
the sounding string. Suppose the microphone, which is actually
intended for the recording of the treble, located one meter further
away from the sounding string, as the microphone for recording the
bass. The sound needs a 1: 340 th of a second longer to treble
microphone. In mixing the signals from both microphones now come
together. In the nature of the sound, both are almost identical,
after all, they have taken up the same sound, but they are
time-shifted slightly against each other. It can be assumed that it
now provides a panorama of the microphone to the left and the other
to the right. Thus, the problem is actually eliminated. Now if a
listener is exactly in the middle of the speakers, which reproduce
the piano sound, it will at least be able to hear in the deep tones
that there are phase cancellation. If you switch now this stereo
signal to mono, the two slightly offset signals add together. Now it
comes to cancellations of individual frequencies. Compared with a
recording of the same instrument with only a microphone, there will
be a lesser amount of bass. How serious are the cancellation depends
on the ratio of the volume of both signals. The greater the
difference in the volume, the less is the problem. If both signals
are approximately identical, arising comb filter effects. The human
ear will perceive such a comb filter effect than a changed tone.
It
is known that the human ear for delay distortions or phase distortion
is much less sensitive than, for example for amplitude distortions.
But the ear may very well perceive the resulting errors. You identify
it only takes a little practice.
To
learn how a phase cancellation effect in a complex signal, one can
tentatively change the polarity on one of his speakers. The membrane
of a loudspeaker vibrates at a resting point. Now, if the polarity is
reversed a box that has the result that the membrane of a box goes to
the outside at a signal and at the other box inwards. This has the
consequence that a box generates a sound-pressure and the other box,
a sound negative pressure.
This
can be heard. One gets the impression as if the bass reproduction to
be considerably weaker.
In
1988 I began in Frankfurt in a recording studio as an assistant,
coffee maker Pizzaholer and studio technicians ,. This studio was set
up for "audio file" Sound Recording. It had a big recording
room with a grand piano and a control room with incorporated
premeasured Studio Acoustic (live end / dead end). We have there
recorded and produced many jazz and rock bands. In receiving free
times I was the studio often for its own sessions and
sound-engineering experiments. Many of my knowledge and experience I
have acquired to me there. When shooting our wing I have tried many
variations on the one hand the best possible way to capture and
secondly to minimize the resulting phase shifts through the
"Polymikrofonie" the sound of the instrument. By the way,
that has my ability to perceive even small phase shifts, trained.
Apart
from the reduced bass playback leaves a phase shift an indefinable,
skuriles feeling in the head. A friend described it as follows: It
feels like someone driving with a sponge over your scalp.
Recently,
I have a plug-in of the company Universal Audio named "Little
Labs" discovered. It offers the possibility of the phase of a
signal of 0 degrees to 180 degrees to rotate continuously, and it can
delay the signal continuously from 0 to 4 milliseconds. With this
tool, you can compensate for the phase problems of a stereo signal
effectively. As a control, I take a phase correlator for help. The
less left of center moves the indicator, the optimal will be the
result.
Possible
technical causes of phase cancellation:
Often
I have XLR cable soldered; An XLR connector has three numbered with
numbers Anschüsse, No. 1 is the connection for the shield, No. 2 for
the positive signal wire, No. 3 for the negative signal wire. When I
2 and 3 Swap connection to one of the two male ends, I create in
conjunction with a second (properly connected cables) a 180 degree
phase cancellation.
Modulation
effects such as chorus produce phase shifts that can cause
cancellations of low frequencies. Therefore, the chorus on a bass is
critical.
Amplitude
distortion
Amplitude
distortion called all changes an original signal, wherein the
amplitudes of individual frequencies are amplified or attenuated.
Are, for example, raised the high frequencies, the sound will be more
brilliant than the original. If the low frequencies increased, you
will feel the sound "more powerful" than the original
sound.
With
an equalizer amplitude distortions are deliberately brought about in
order to make a sound in a certain way.
In
the German translation the word Equalizer "leveler" means.
Originally
an equalizer was used in the broadcast technology to compensate for
unwanted amplitude distortions of a transmission chain. It should
therefore be "razed" by means of an equalizer, the
transmitted sound to the original.
There
are a number of polluters of amplitude distortion.
Each
cable amplitude distortions produced by the electrical resistance of
the copper wire. If all frequencies equally reduced, you will be able
to perceive the change only in the volume. Will some frequencies more
reduced than others, one perceives a change of tone. The problem is
known in a guitar cable. Since the output of an electric guitar is
high-impedance, the cable acts as a filter that reduces high
frequencies.
Amplitude
distortions also arise because an amplifier not all frequencies of
the audio spectrum accurately amplifies the same. Especially at very
low and very high frequencies that is relevant. A quality
characteristic of amplifier is the frequency response (20-20,000 Hz
for example)
This
means that the amplifier is capable of all frequencies between 20 and
20,000 Hz with (almost) the same volume can be transferred.
Relatively
strong arise unwanted amplitude distortion in a loudspeaker or a
loudspeaker. In most chains About Conference speakers are the
"weakest" element, because it causes the most massive
linear distortions.
Nonlinear
Distortion
Includes
a sound non-linear distortions, the original have been added
intentionally or unintentionally overtones. Most is integral
multiples of the fundamental frequency. The ratio of the root of the
unwanted overtones is called distortion. The louder the harmonics
relative to the fundamental, the higher is the THD. These distortions
are caused by the non-linear characteristics in the active components
(transistors, ICs, amplifier tubes) of an amplifier. So there are the
non-linear distortions, which are referred to in common parlance as
distortions. Due to the high quality of today's electronic
components, one can assume that you have this kind of distortion only
perceives when overloaded an amplifier massively. The resulting
congestion in the heat also plays a role, because the temperature
often influence on the characteristic of an active component.
The
most famous use of non-linear distortions, there is in the world of
guitarists. What is usually unintentionally, is looking for a
guitarist. Originally, the distortion created only if you hired a
guitar amplifier for maximum gain and thus overburdened. The
guitarist enjoys not only the overtones newfound, but he also uses
the fact that the Sustain extended by the distortion, or in other
words, that the sound is longer before he stopped. Since this effect
is desired to produce only in connection with a brutal volume, the
industry has been built since the 60s devices that generate at least
this or a similar distortion at low volume. They call themselves
Fuzz, Tube Screamer, overdrive or distortion.
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