Generate loud, rich sound from micro speakers

Generate loud, rich sound from micro
Know the challenges that hinder improvements in audio performance, and how to
overcome them.
By Shawn Scarlett
Director of Marketing
Mobile Audio
NXP Semiconductors
While the video screens of mobile phones, tablets and notebooks have seen astonishing improvements, audio
performance has lagged far behind. Phone speakers still sound quiet and tinny, limited by their tiny size. Designers
use various techniques to increase the volume and sound quality, but with limited success. They also bring risks:
blown speakers are a common cause of failures in mobiles.
Simply limiting the output power makes for a poor user experience, and doesn't protect against blocked speaker
ports or high ambient temperatures. Temperature measurements can help but do little to improve sound quality.
High-pass filters reduce the speaker excursion at the resonant frequency but cut out too much bass.
Feed-forward techniques can improve bass response but on their own aren't enough and the can be a reliability
risk. Additionally, clipping and low battery voltages can degrade sound quality even further.
Speakers come full circle
Speakers and phones have developed hand-in-hand for over 150 years. The first speakers were used in telephone
receivers, shortly afterwards they branched off into sound reinforcement and grew larger and more powerful.
In the 1980s and 90s things came full circle. Modern mobile phones have two speakers. One, still called a receiver, is
in the earpiece. The second is for sound reinforcement, for things like ringtones, music playback and hands-free
Micro speakers try to bridge the gap, aiming to produce room-filling sound from a tiny volume. What began with a
move to play better polyphonic ringtones has now grown toward using a cellphone instead of a home stereo. These
speakers are caught between two opposing trends, more output power and smaller size. As these trends accelerate,
speaker designers are starting to look for new and innovative ways to get the best possible sound.
Modern micro speakers have a permanent magnet and a voice coil that is attached to a diaphragm that pushes the
air to create sound. The entire speaker is enclosed in protective box that provides the "back volume" for the speaker
to push against and project the sound from the speaker.
Output limited by temperature
The first way to get more sound out of a speaker is simply to put more electrical power in. Small micro speakers
rated at ½ Watt can generally handle many times that for very short periods. All the extra power going in has to
come out somewhere, though.
Figure 1: Dissipating too much heat can tear the voice coil apart.
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Maximizing efficiency converts as much power as possible into sound. However, much is still wasted as heat in the
voice coil. This 'self heating' is directly related to the current in the voice coil. If the temperature climbs too high, the
glue holding the voice coil together can be torn apart (figure 1).
The speaker is cooled by conducting the heat out through the membrane, case and other components and by the
cooling effect of moving air from the sound waves themselves. Lower frequencies generate more air movement
causing more cooling and hence allowing higher powers.
This relation breaks down if the speaker port is blocked, the air movement is restricted or the ambient temperature
rises. If the air cannot cool the coil, the internal temperature rises much faster than expected, and the speaker can
be damaged in a few seconds. The relationship between coil temperature, power level, frequency, duration, ambient
temperature, and airflow is complex, and is virtually impossible to reliably predict.
Speaker excursion
Because micro speakers must be small, it is easy to move the diaphragm further than the maximum allowable
excursion (typically around 0.4 mm). As speakers get thinner, the excursion becomes smaller, which is a major
restriction on output sound level.
A speaker's biggest excursion problem comes at and near its resonant frequency. At the resonant frequency the
membrane moves easily, so small amounts of power can push the speaker beyond its limit. Micro speaker systems
normally add a high-pass filter at around 1000Hz to reduce the excursion. This can minimize the impact of the
resonance peak, but losing the bass significantly degrades the sound quality.
The resonant frequency can change dramatically over the operating conditions, too. Temperature, aging, a poorly
designed phone case, and changes in the acoustic environment like blocking a speaker port will all cause shifts in
the resonant frequency. Wear-and-tear on the phone case can also cause leaks in the speaker's back-volume. Any of
these changes can cause speaker failure in a fixed-filter system.
Clipping and power supply sag
A 2 VRMS power level corresponds to a 2.8 V peak amplifier output. Nearing this level causes clipping and
distortion. For higher frequencies, where excursion is not a problem, this is the main issue affecting sound quality.
It is worse with power supply 'sag' because of low battery voltage or high current drain. Since sag is often caused by
the audio amplifier itself, this is hard to solve. Some systems lower the amplifier gain when the supply sags however these cannot typically react fast enough for dynamic signals and still distort the peaks.
Boosting the supply level with integrated DC/DC converters can reduce amplifier clipping by adding headroom, but
system designers must be careful not to damage the speaker by over-excursion. The voltage boost can also increase
peak battery current and cause the voltage level of a partially discharged battery to drop low enough to cause a
system reset, resulting in a dropped call or audio glitch.
Figure 2: Safe operating area for a speaker.
Safe operating range for micro speakers without protection
All these parameters can delimit an area of safe operation (figure 2). A temperature line limits the power amplifier
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to avoid the worst-case self-heating temperature, and a frequency line removes frequencies below resonance to
prevent over-excursion.
Allowing for changes in the acoustic environment and ambient temperature ensures safe operation, but with only a
modest sound output.
For a typical system, limiting the power input to the speaker to 2 VRMS (2.8 Vpeak) and adding a 1kHz high-pass
filter will create a system that remains inside these limits. When used with an 8 ohm speaker it will result in less
than 0.5 W (0.9 W peak) output.
Improving volume output
Systems should always operate near peak output. Because audio signals are dynamic, they only rarely use the
amplifier's peak output voltage. Compressing the signal's dynamic range (Figure 3) increases the apparent volume
without changing the peak levels (which is why commercials sound louder than the rest of the TV or radio
These dynamic compressors work by adding gain to the quiet parts of the music, and quickly reducing it at peaks
(the 'attack time'). The attack time is typically very fast (50 µs) and the corresponding decay time over which the
gain is increased is typically much slower (5 seconds).
This approach again brings risks. Peak audio signals near the resonant frequency can see very large gain within the
attack time. This increases the potential for over excursion and damaged speakers.
Figure 3: Sound sample before and after compression.
The output volume can also be increased by filtering out the resonant frequency. By removing the frequencies near
resonance more power can be applied to the remaining signal. That drives more sound from the speaker, but the
missing frequencies degrade sound quality.
The filter can be improved and narrowed by using models to predict the behavior of the resonant frequency and
speaker temperature. However, any mismatch between the model and the real world can be catastrophic. A blocked
speaker port, for example, changes the resonant frequency, with the filter then not protecting the speaker from
Predictive models in these feed-forward systems can also calculate the speaker excursion based on the input signal.
That can allow some frequencies below resonance back into the signal which improves sound quality, but it also
compounds the risk, because high power signals can be delivered to the speaker where it is vulnerable to damage.
The feedback solution
Eliminating the differences between such complex models and the real world requires feedback. Feedback systems
use real-world measurements to update the internal models that predict speaker behavior, and allow the system to
produce more sound safely.
Key is to directly monitor the voltage and current to the speaker. This is not as easy as it sounds, since most
portable audio systems use class-D amplifiers to reduce power consumption. The sample must therefore be taken
after the signal is converted back to analog, which means using an external sense resistor after a power filter. This
resistor lowers the system efficiency, because it consumes some of the output power.
Alternatively, more advanced current-sensing systems can be synchronized with the amplifier switching. This
approach can provide more accurate results for small systems that don't use power filters on the amplifier output.
This solution can be fully integrated inside the amplifier, reducing output pins.
The first step in a feedback system is to measure the speaker voice coil temperature. Because coil impedance rises
linearly with temperature, an accurate current measurement can provide a stable and accurate temperature
measurement. This can accurately protect against thermal speaker overload as long as manufacturing variations are
properly accounted for during production.
The next major step in protecting the speaker comes from controlling the excursion directly. Basic feed-forward
systems can measure temperature to estimate the speaker resonance. More advanced systems use current sensing
to accurately measure the impedance across all frequencies. The impedance spectra generate an adaptive model
which can accurately predict the speaker excursion.
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With direct information on excursion, a system can always drive low frequencies into the speaker without damage.
If the speaker port becomes blocked, the resonant frequency changes and the system will adjust the signal to
prevent damage.
The excursion information can also be used to optimize the output from the speaker, rather than optimizing a fixed
electrical level. Here, the speaker can always use the maximum possible excursion for the desired signal. That also
improves the sound quality by avoiding clearly audible distortion caused when the speaker moves beyond its limits.
Feedback can also use information from the DC/DC converter to optimize sound quality and system performance.
Monitoring the current and voltage at the DC/DC converter can detect supply sag and adjust the peak output
accordingly. This can ensure that the audio signal is never clipped, and sound quality (along with system
performance more generally) will not degrade as the battery discharges.
Additional feedback points can further improve sound quality and system performance while also avoiding the risks
of using higher supply voltages. This brings a huge performance improvement in SPL, sound quality and speaker
A feedback-based solution gives several key advantages by automatically adapting to changes in acoustic and
thermal environments. A full solution, however, must use a combination of techniques.
Adaptive excursion control is needed to ensure that the speaker membrane excursion never exceeds its rated limit.
Real-time temperature protection is needed to directly measure voice-coil temperature to prevent thermal damage.
A design must prevent clipping even with sagging supply voltage, and bandwidth extension must increase the low
frequency response well below speaker resonance. And an intelligent DC-to-DC converter is needed to maximize
audio headroom even at low battery voltages.
Benefits of additional DSP
System designers would prefer to integrate any processing into the main system. Processing in the phone chipset
will generally indeed give the smallest, most power efficient, and cheapest solution. However, accurate feedback is
the key to successful speaker boost, and it needs low latency and high bandwidth.
Comprehensive speaker protection actually requires multiple input points - just knowing current and voltage isn't
enough. Furthermore, interrupts and system integration issues can become a major hassle. Multiple sensing points
are needed to optimize the amplified signal. The processing must also optimize the performance of both amplifier
and DC/DC converter.
To properly integrate this system into the central processor, all these signals would need to be converted and fed in
to the chipset and all the controls properly taken out. A separate DSP can handle all these interactions automatically
and can run continually even when the central processing shuts down.
Speaker boost and protection
NXP's TFA9887 is touted to boost output while protecting the speaker. It has an embedded CoolFlux DSP, Class-D
amplifier with integrated current sensing, and intelligent DC/DC boost converter.
The IC holds a software model of the speaker, and automatically adapts to any changes over the speaker's lifetime
including aging, enclosure damage, blocked speaker ports, or whatever the world can throw at it. Better sound
quality can also be traded against even smaller speakers and back volumes, giving smaller end products.
To confirm its performance, we compared the SPL of a speaker driven by the TFA9887 with a popular unmodified
smart phone which uses a software compressor to enhance the volume. The test used identical test files and an
identical phone (so identical speaker and enclosure).
Figure 4 shows more than 6 dB SPL increase in output volume. Optimized for bandwidth, bass output is increased
by around 10 dB SPL.
Figure 4: Volume and bass levels from the TFA9887.
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This kind of performance illustrates an important design trend. The days of standalone amplifiers and converters
designed in isolation have gone. The performance of phones and other portable devices have seen so many
refinements that components must be treated as part of a bigger system. Each part of the system must sense and
interact with the real world for best possible system performance.
So, audio systems must monitor the performance of the acoustics and adjust for the best user experience. Here as
elsewhere, there is a clear trend to producing systems that measure and interact with the real world. About the author
Shawn Scarlett is the marketing director for NXP’s mobile audio group, with a long history specializing in audio
semiconductors including positions at Analog Devices and National Semiconductor, as well as startups such as
Tripath and GTRonix. He has a B.S. in Electrical Engineering from the University of Arizona as well as an MBA from
Santa Clara University. Before moving into semiconductors, he developed his audio skills as a professional sound
engineer with the I.A.T.S.E working on sound reinforcement for major touring shows.
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