S5 SOUND PROCESSORS

5 SOUND PROCESSORS.

5.1 LEVEL OF SIGNAL AND MEASUREMENT UNITS.

5.2 AMPLIFIERS.

5.3 FILTERS AND EQUALIZERS.

5.4 DYNAMICS PROCESSORS.

5.5 EFFECTS PROCESSORS.

5.6 SURROUND PROCESSORS.

5.7 ADAPTERS AND PHONES ENCODERS.

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5 SOUND PROCESSORS.
As shown eairlier in this lesson, it has been taken into account all
considerations to deliver the audio signal from a source such as a
microphone to other equipment through transmission lines in optimal
conditions. At the end of these lines we can find several simple or mixed
process like:
• Signal amplification

• Change in one of its parameters as (frequency, time, period, etc..)

• Addition of different signals

• Subtraction of signals
• Effects.

According to that, It is consider a audio processor those equipments that

modify the signal audio characteristics without compromising its harmonic
integrity. By contrast are consider audio effects all those that change
 5 SOUND PROCESSORS
5.1 LEVEL OF SIGNAL AND MEASUREMENT UNITS.
The input and output signals of any sound equipment have different values, so
before starting to study the different processing equipment, it is necessary to know
the different types of signals and their units, so that we can classify, measure,
quantify them and not make mistakes when interconnecting different equipment
that require specific types of signals and units.:
5.1.1 SIGNAL TYPES
• Microphone signal: The main characteristics of this type of signal are that they have
low impedance and provide voltage levels on the order of millivolts (for example, 150Ω
and 2mV). They typically require a preamplifier to be raised to line level and be able to
be processed by the following equipment in the audio chain.
• Line signal: The primary characteristics of this type of signal are that they have a
medium impedance level and provide signal levels of +4dBu in professional
environments and -10dBV in domestic environments, meaning signals in the hundreds
of millivolts (e.g., 1mV and 1V). This is the most versatile signal that most equipment
and processors can handle.
• Load signal: It is the signal that the amplifier provides to a loudspeaker. These types
of signals are characterized by having very low impedance and high voltages, which
depend on the amplifier's power and the gain levels selected on it. As example values,
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5.2.1 SIGNAL TYPES
• Instrument signal: These are the signals produced by electric instruments
such as electric guitars, basses, or keyboards. These signals typically have
voltage levels around 1 volt, which classifies them as line-level signals. They are
high impedance and require impedance matching through injection boxes for
connection to other equipment. Injection boxes provide a microphone-level
output. Alternatively, equipment can have dedicated inputs for these types of
instruments, capable of handling signals in the range of 1V and 15KΩ ohms.
• Record signal: It's a medium impedance signal that typically provides 2mV for
every 1KΩ of input impedance of the recording equipment. Since the input
impedances of recording equipment are usually in the range of 50 to 100 KΩ,
the output voltages are typically in the range of 100mV to 200mV. Dani,
remember to explain this relationship using the appropriate rule of three.
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5.1.2 MEASUREMENTS UNITS
• dBu: It is a logarithmic unit of voltage measurement, with a reference voltage
of 0.775Vrms (under conditions where this voltage is applied to a 600Ω resistor
and dissipates 1mW). It is primarily used in professional audio environments.

dBu=20log(Vmeasurment/Vref)=20log(Vmeasurment/0,775)

• dBV: It is a logarithmic unit of voltage measurement, with a reference voltage

of 1Vrms. It is primarily used in home audio environments.

dBV=20log(Vmeasurment/Vref)=20log(Vmeasurment/1)

• Vrms: Root Mean Square voltage or effective voltages is a statistical measure of

the magnitude of a varying voltage. It is calculated using the following formula.

Vrms=Vpico/√2

• dBm: It´s a logarithmic unit of power meassuremedt, with a reference power of

1mW (under conditions where this power is applied to a 600Ω resistor). It is
primarily used in professional audio environments, where dBm is coinciding with
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5.1.2 MEASUREMENTS UNITS
• dBm: It´s a logarithmic unit of power meassuremedt, with a reference power of
1mW (under conditions where this power is applied to a 600Ω resistor). It is
primarily used in professional audio environments, where dBm is coinciding with
dBu.

dBm=10log(Wmeasurment/Wref)=10log(Wmeasurment/0,001)
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5.1.3 SUMMARY SIGNAL AND MEASUREMENT UNITS.

Nivel de carga
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5.2 AMPLIFIERS.
They are responsible for increasing signal levels, either in
voltage or current, to adapt the audio signal to the next
stage that needs to be connected, within the sound
chain. Depending on their position in the sound chain,
they may have the function of adapting levels between
stages, which do not require large amplification ranges,
or on the contrary, be part of the final stage before the
speakers where large amplification ranges and power are
normally required. Their electronic architecture is
composed mainly of transformers and transistors that
consume a lot of current and consequently generate a lot
of heat, which makes them equipment that must be
properly ventilated.

The parameters that characterize an amplifier and

help us to compare different brands or equipment in
order to make a good choice are:
• Gain: relation between output power (Po) and
power in (Pin) usually expressed in dB
G=10xlog(Po / Pin)
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5.2 AMPLIFIERS.
• Power Po: It is the electrical power that could be applied to the speakers
without producing distortion and without damaging the equipment when
connected to a nominal impedance.
• Frequency response: It represents the relationship between the input and
output signals as a function of frequency. Ideally, this response should be flat
from 20 to 20KHz.
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5.2 AMPLIFIERS.
• Frequency response
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5.2 AMPLIFIERS.
• Distorsion THD(Total Harmonic Distorsión):
Distortion in an audio amplifier is the deviation of the
output waveform from a precise replica of the input
waveform, often caused by nonlinearities in the
amplifiers components or circuit design. The main
effect is the generation of new frequency components
that are integer multiples of the fundamental frequency
of the input signal or when an amplifier is driven
beyond its maximum output capability causing the
clipping.
• SNR: Difference between signal output and noise generated by the amplifier.
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5.2 AMPLIFIERS.
• Impedance (Z): As we Know, impedance is the opposition a circuit offers to the
flow of alternating current. In sound amplifiers, impedance is crucial in order to
obtain a maximum transmission power from in to out and get a god impedance
adaptation. For impedance we have to differentiate between Zin and Zout:
o Zin : It´s the opposition an amplifier offers to the signal source connected to its input.
It's essentially the load seen by the preceding stage (like a preamp or a microphone).
It is generally desirable to have a high input impedance to minimizes loading effects
on the source, ensuring that the amplifier takes as little signal as possible from the
preceding stage and allows for a wider range of devices to be connected to the
amplifier without significant signal loss. This results in a more accurate reproduction
of the original signal.
o Zout : It´s the opposition an amplifier presents to the load (usually speakers) connected
to its output. A low output impedance is typically preferred as it allows for better
power transfer to the speakers, resulting in louder and clearer sound. It also helps to
ensure a more stable and consistent performance, even when driving different
speaker loads.
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5.2 AMPLIFIERS.
• Crosstalk: It´s very common that an amplifier have more than one power
stage/channel/via 2, 4, 6 or even more. In this cases it could be possible that
part of the signal of a channel can mix in other channel and this is Knew as
Crosstalk. Channel separation is the inverse of crosstalk, and quantify the
attenuation between channels. Higher values indicate superior channel
separation and reduced crosstalk. Typically, values exceeding 30 or 40 dB are
considered indicative of good channel separation.

DANI EXPLICA ESTO CON LA MESA DE MEZCLA

CONECTA DOS MICROFONOS Y PON EL BALANCEO
DE UNO A LA IZQ Y EL OTRO A LA DRCH.
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5.3 FILTERS AND EQUALIZERS.
The function of these equipments is to modify the original frequency response of a
circuit through which an audio signal passes. This type of procesing affect to the
signal level as a function of frequecny, and they are used in order to fix technical
failures or for artirstic purposes. The most characterize applications for this
processor are:
• Room equalization
• Sound reinforcement
• Correction of faults caused by sound connections or signal recording.
5.3.1 FILTERS.
These processors discriminate frequencies and allow a certain margin of them to pass through
while blocking the rest of the frequencies found within that margin. Depending on the type of
electronic components used for their design and construction, filters can be active or passive.
In the case of passive filters, the components used for their implementation are coils and
capacitors, and their operating principle is based on the variation of the impedance of these
components as a function of frequency. In general, filters are classified as follow:
• Low pass filter (LPF): allow frequencies below the cutoff frequency to pass, blocking
upper frequencies.
• High pass filter (HPF): allow frequencies above the cutoff frequency to pass while
attenuating lower frequencies
• Band pass filter (BPF): It´s a combination of HPF and LPF, allowing frequencies
between its cutoff frequencies and blocking the rest.
• Band stop filter (BSF): Block frequencies between its cutoff frequencies. If the cutoff
frequencies are very close are Knew as notch-filter. They are often employed to remove
narrowband interference.
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5.3.1 FILTERS.
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5.3.1 FILTERS.
Apart of by its cutoff frequencies, that are defined as the frequencies when signal is
attenuated 3dB. A filter is characterize by these parameters:
• Central frequency: This parameter is use en BPF and BSF and is defined
acording to follow mathematics expression.

Fc=√(fcs x fci)

Remember fc isn´t = (fcs-fci)/2

• Bandwith: It´s the margin of frequencies a filter allow to passs or block and it´s
limited by the cutoof frequencies.

BW3dB= fcs - fci

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5.3.1 FILTERS.
• Filter order: It numerically represents the drop slope or the attenuation of a
filter and is obtained as a result of the difference in attenuation per octave
between 6. Remember that an octave is characterized by the upper frequency
being twice the lower frequency, and it should not be confused with cutoff
frequencies. The higher the order of the filter, the greater the attenuation slope.
The order of passive filters depends on the number of passive components; for
example, a first-order filter (implemented with a capacitor or an inductor)
decays at 6 dB/octave.
Nº of order=attenuation per octave/6 dB
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5.3.1 FILTERS.
• Quality factor: It allows quantifying the selectivity of a filter in a way different
from its order. It is obtained by dividing the filter's center frequency by its
bandwidth. The higher its value, the more selective the filter. It is primarily used
to characterize notch filters.
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5.3.2 EQUALIZERS.
These devices apply a certain gain or attenuation to a set of frequencies, but do not
block any. They serve to emphasize desired frequency groups or attenuate those that
are annoying or inadequate. In general terms, these equipments can be classified as
follows:
• Graphic equalizer: It consists of a set of filters connected in parallel, covering
the entire audible spectrum, allowing modification of the frequency response of
any equipment within the sound chain. The number of band depend of the use,
a few for domestic environments until 31 in professional environments using
one-third octave filter. The attenuation or gain range is 12dB per band, and its
primary use is for final equalization of a mix and adjustment to correct room
acoustics issues.

• Parametric equalizer: Unlike graphic equalizers, which have their center

frequencies defined by ISO standards, parametric equalizers allow for variation
in both the center frequency and Q factor of their filters, making them fully
customizable equalizers.
• Semiparametric equalizer: It´s a parametric equalizer without the option to
set-up the Q factor.
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5.3.2 EQUALIZERS.
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5.4 DYNAMICS PROCESSORS.

The dynamic range of a equipment is defined as difference in dB, between
the noise level and the maximum admissible level. It is desirable for the
dynamic range to be large, so that equipment can handle signals that
fluctuate between high peak values and subtle signal values.

The audio signal can come from different sources, a instrument, a voice or
song mixed that can have different dynamic range and can be modify for:
• Solve the lack of dynamic range in specifics transmission system as wireless
microphones.
• Reduce the noise and improve the dynamic range.
• Controlling excessive levels of an audio signal.
• Increase the sonority of an audio program.
• Generate new different feelings of an instrument in mix.
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5.4 DYNAMICS PROCESSORS.

These audio devices can be analog or digital and modify the dynamic of
audio signal, that is, the difference between their maximum and minimum
levels. It can be classified into the different types shown in next slides.
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5.4.1 COMPRESSORS AND LIMITERS.

The compressor is the most dynamic processor used, and its function is attenuate the
input signal when a certain threshold level is exceeded. In this way, the signal peaks
are attenuated and the general signal dynamic will be lower. A typical parameter of
these equipments is the compression ratio that indicate how many dB are attenuated
for each dB that exceed the threshold, for example a ratio 2:1 indicate that for each dB
that exceed the threshold, the input signal will be attenuated 2 dB.
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5.4.1 COMPRESSORS AND LIMITERS.

The limiter is a especific case of a compressor. Instead of introduece a certain level
of attenuation to the input signal when a certain threshold level is exceeded, they
attenuate the signal until the threshold level. The compression ratio of these
equipments is ∞:1

The compressors are used to compress the dynamic of bass, guitars, drums and voices,
meanwhile the limiters are used to avoid the music exceed dB levels established by
authorities in bars, clubs and nigth clubs.
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5.4.2 EXPANDER AND NOISE GATE.

Expander makes the inverse function of a compressor. These equipements attenuate
the signal that is below of a threshold without attenuate anything above the threshold.
These equiments are usefull when it is necessary improve the SNR, and wit
technologies as cassetes and disc playesr where the dynamic range is not enough.
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5.4.2 EXPANDER AND NOISE GATE.

The noise gates work as an expander but with an attenuation ratio of 1:∞, It means
that any signal below the threshold is completely attenuated. These equipments are
frecuently used to isolate the useful signal from the noise of other sources, for instance,
on the microphones that capture the sound of the bass drum or snare drum of drum kit
concert.
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5.5 EFFECTS PROCESSORS.

These equipments mainly affect the signals timing, applying delays,
modulation, and addition to the original signal. These effects can also affect
to the frequency distribution similar to an equalizer. Nowadays this process
is made with DSP (digital signal processor), Among the more common effects
we can find the following:
• Reverb: they are effects that simulate the multiple natural reflections
that occur in an enclosure (concert hall, cathedral, bathroom, etc.) and
provide greater naturalness to certain sounds such as voices or
guitars. Reverb equipment allows control over the distribution of
reflections in time, their intensity, and their frequency response.

• Delay: these devices store the input signal for a specific time according
to their configuration, and then reproduce it at their output with a
delay. They allow the simulation of echoes and the adjustment of
sound systems in large spaces such as cinemas or theaters.
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5.5 EFFECTS PROCESSORS.

• Coro (Chorus): It consists of a delay of a few milliseconds that is added
and modulated to the original signal, creating a sound effect of
multiple performers. This effect is frequently applied to guitars, basses,
and keyboards.
• Flanger and phaser: These effects are created by mixing two signals:
the original and a slightly delayed version. The resulting phase
cancellation produces a distinctive, unnatural sound.
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5.6 SURROUND PROCESSORS.

These systems employ multiple loudspeakers to produce a spacious and immersive
soundstage. The speakers are arranged around the listener to produce an immersive
sound field that replicates the acoustic characteristics of the original recording space.
Although there are different surround sound standards, they all share the following
common channels, which we will now detail.
• Right and left channels: the left and right channels correspond to the
traditional stereo channels. They are located in front of the viewer, on the sides
of the enclosure. In multi-channel systems, these channels are dedicated to
transmitting the main ambient sounds and the music of the audiovisual
production.
• Central channel: the center channel is located in front of the viewer and
centered according to their viewing position. It is primarily dedicated to voice,
that is, to conversations when watching a movie or to singers when watching a
concert.
• Surround channels: It could be one or several channels dedicated to ambient
effects that will recreate a realistic sensation of sound immersion in audiovisual
production. If it is a single effects channel, it will be called surround, and if there
are two, they can be called left surround (SL) and right surround (SR)
• Subwoofer channel (LFE):To produce realistic and impactful low-frequency
sounds, such as explosions or earthquakes, a specific channel is used that
carries low-frequency signals to large speakers capable of producing sufficient
sound pressure level and vibration. Due to the omnidirectionality of these
frequencies, a single channel is sufficient to cover the listening area of the room.
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5.6 SURROUND PROCESSORS.

Considering the number of channels and their position in the room, we can make
the following classification:
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5.6 SURROUND PROCESSORS.

Considering the standards, we can make the following classification:
• Dolby Digital: As one of the pioneers in surround sound, Dolby Digital
has a long-standing reputation for delivering reliable and high-quality
audio experiences. Its compatibility with a wide range of devices
makes it a popular choice for home theaters.
• DTS: Known for its advanced audio coding, DTS offers a competitive
alternative to Dolby Digital. With a focus on dynamic range and
precision, DTS delivers a cinematic audio experience.
• Dolby Atmos: revolutionized home theater audio by introducing the
concept of audio objects. This technology allows sound to be placed
precisely in a three-dimensional space, creating a more realistic and
immersive listening experience.
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5.6 SURROUND PROCESSORS.

• DTS:X: it is a direct competitor to Dolby Atmos, offering similar object-
based audio capabilities. Both formats provide a highly immersive and
customizable surround sound experience.

• Auro-3D: it is another leading player in the object-based audio

market. By placing a strong emphasis on the height channel, Auro-3D
creates a more enveloping and realistic soundscape, making listeners
feel like they are in the center of the action.
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5.6 SURROUND PROCESSORS.

• Auro-3D: it is another leading player in the object-based audio
market. By placing a strong emphasis on the height channel, Auro-3D
creates a more enveloping and realistic soundscape, making listeners
feel like they are in the center of the action.
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5.7 ADAPTERS AND PHONES ENCODERS.

In radio and television productions, it is common to include signals from
telephone lines in the sound mix. This occurs when connecting with a
correspondent or when interviewing a guest by phone on a news program.
Landline and mobile phone systems are not designed to be connected to audio
systems, and therefore it is necessary to have specific equipment that allows for
the bidirectional extraction, insertion, and recoding of audio. This equipment is
called telephone hybrids.
Telephone hybrids connect to both a phone line and a telephone terminal for call
management. The interlocutor's voice is routed through a dedicated audio output
to a mixer input, like a microphone. To enable the caller to hear studio audio, a
mixer output is connected to the hybrid's return input.
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CHIMPUM