FIR Filters Help Improve Audio Quality in Consumer Electronics
Consumer Audio Devices from TVs to Docking Stations to Personal Computers
Benefit from the Use of FIR Filters in Providing High Quality Audio at a
Reasonable Cost
By QuickFilter Technologies, Inc.
The U.S. market for consumer electronics goods is approximately $161B and
growing at a rate of 8% per year (Consumer Electronics Association Digital
America 2008). Television remained the biggest driver fueled by the growth of
LCD TVs at 32% CAGR and Portable audio devices with a CAGR of 22%. To date, the
key product differentiator has been unit price. As the market matures,
differentiation will proceed along two vectors; one being unit price and the
other being audio/video quality. In the past, these have been thought to have
been mutually exclusive. Great strides have been made in addressing video
quality, but audio quality has continued to be subpar in Digital TV and
portable audio devices.
click to enlarge
Figure 1. Audio precision measurement of speaker transfer function. |
Audio quality has often been sacrificed in order to achieve the cost structure
necessary to be competitive, but as the market matures, audio quality will
become an important product differentiator. The compromise in audio quality is
often reflected by the manufacturers' choice of speakers. The speakers chosen
by these manufacturers are constrained by their cost, the physical size of the
equipment, and the manufacturing process in mounting the speakers. In making
the design and cost tradeoffs, little attention is given to the speaker
response over the audible frequency range (speaker transfer function) because
it is thought that the equalizers in the audio processors will compensate for
flaws in these lower cost speakers. This rarely happens. Listen to the
soundtrack of a DVD played through a $3,000 HDTV and the same soundtrack played
through a good quality sound system. The difference is very noticeable. With
some minor changes, however, manufacturers can approach the audio quality of a
high end sound system with very little to no incremental cost.
Many manufacturers believe that good audio quality is synonymous with maximum
bass; however, audio quality is really the ability to reproduce the sound track
exactly as it was recorded. To achieve this, the speaker response needs to be
flat across the audible spectrum. Manufacturers of high-end stereo equipment
have recognized ? and achieved ? this requirement. In many cases, these
manufacturers ship their equipment with no bass or treble adjustments.
Manufacturers of consumer electronics equipment can achieve this same
performance by using Finite Impulse Response (FIR) filters that are low in cost
but high in performance. Typical audio devices which will benefit from the use
of FIR filters include:
-
Televisions, especially high definition TVs
-
Docking stations
-
Mid to low end stereo systems
-
Boomboxes
-
Headsets both wired and wireless
-
Personal computer audio equipment
An ideal speaker would be capable of reproducing all the audible frequencies at
the same volume at which they were recorded. Rarely in life are things perfect,
so the generally accepted standard is a variation of ±3 dB around a
centerline. The 3 dB point is considered to be the point at which volume
changes are barely perceptible. Even if a speaker transducer were to possess
this characteristic, the process of mounting that speaker into a cabinet will
change its response characteristics, so compensation will still need to be
applied after the speaker is mounted in its final enclosure. The use of an FIR
filter will simplify this compensation.
The graph shown in Figure 1 was taken with an Audio Precision 2700. It measures
the frequency response of a name brand 40 inch LCD HDTV. It is clear that the
response is not flat. The variation around the centerline (of-18) varies from 6
to 8 dB. The lower frequencies are overemphasized and tend to wash out the mid-
to high-end ranges. Also, notice the significant drop in mid-range frequencies.
The mid-range drop will affect the intelligibility of the dialog (particularly
female speech). The goal is to bring response back so that it is within
±3 dB of centerline to ensure the frequencies are heard at the same
volume at which they were recorded.
There are two methods that can be employed to equalize this speaker. The more
traditional way is to use a five-band parametric equalizer comprised of five
Infinite Impulse Response (IIR) digital filters. These are typically found in
today's audio processors. There are two disadvantages to these filters. The
first disadvantage is that they are relatively narrow band and must be placed
strategically to cover the frequencies of interest. In addition, there are
discontinuities at the overlaps which can cause distortion. The second
disadvantage is that IIRs introduce phase distortion (different frequencies are
delayed through the filter by different amounts of time). This again causes
distortion to be heard.
click to enlarge
Figure 2. Docking station block diagram using the QF1DaF12. |
An alternate method to achieving a flat response is to create an inverse of the
transfer function and program it into a digital FIR filter. An FIR filter is
broadband in nature and has absolutely no phase distortion. In theory, if you
inverted every point of the graph in Figure 1 and reran the original test with
the filter in series with the speaker, the resulting response would be a flat
line across the measurement range. In reality, absolute perfection would
require a filter with an extremely large number of taps which translates into
very large die size (and thus high cost) as well as add significant delays into
the audio stream. However, a 512-tap digital FIR filter will have a delay of
<6 ms which is generally considered acceptable. Also, the resolution of this
512 tap filter is approximately 24Hz which means that it can perform reasonably
down to about 100 Hz. This is a very cost effective filter and will yield a
result that, while not perfectly flat, will meet the criteria of being within
±3 dB of a center line reference.
An FIR filter by its nature is designed to attenuate signals. In an audio
application, this attenuation has the effect of reducing the dynamic range of
the audio signal. The loss of dynamic range compresses the volume heard
throughout the passage making the differences between soft passages and
crescendos less dramatic with the overall effect being a less pleasing
listening experience. In the past, to overcome this, the system designer would
have to add a digital gain stage to make up for loss of dynamic range, thus
adding cost and complexity to the design. With the introduction of the QF1Da512
(Audio SavFIRe) from Quickfilter Technologies, this is no longer the case. The
Audio SavFIRe is the industry's first dedicated FIR filter to contain a gain
and compressor stage that directly addresses the issue of dynamic range loss.
The device supports up to 512 taps with 32 bit coefficients and user defined
data word width from 12 to 24 bits, providing the user with superior audio
quality. The Audio SavFIRe processes the equivalent of 50 million MAC
instructions per second and supports sample rates from 1 Hz to 500 KHz, placing
it in the class of mid-range Digital Signal Processors and exceeding the
performance of most audio processors.
The QF1Da512 (Audio SavFIRe) costs less than $1.00 in volume, comes packaged in
a 3 × 3 mm QFN and consumes only 2.8 mW of power providing the user with
a clear price-performance advantage. The Digital Gain and Compression (DGC)
block in the QF1Da512 (Audio SavFIRe) combines both gain and compression in one
operational block. The compression is performed sample by sample with an
instant attack and release. The slope of the compression curve is adjusted as
the gain increases. The gain block is configured to provide up to 24 dB of
positive digital gain. This combines to yield the maximum in dynamic range
without distortion.
Using the Audio SavFIRe, a high-quality mp3 docking station can be designed
with just a stereo codec, a microcontroller, and a simple power amplifier (see
Figure 2.)
For an HDTV sound system, the system designer can use the Audio SavFIRe in
co-processor mode, by placing it between the digital output and the digital
input of the audio processor. Alternately (Figure 4), the HDTV manufacturer
could exchange the Audio Processor for a high-quality (lower-cost) ADC.
Coupling that with the Audio SavFIRe and a digital power amplifier would yield
a digital audio subsystem of exceptional quality at lower cost than their
current audio subsystems.
With this impressive new integrated circuit, manufacturers of consumer audio
equipment can now take a significant step on the price performance curve.
H1 Eliminating Background Noise – The Next Best Audio Quality Improvement
By Lloyd Watts, PhD, Chairman, Founder & Chief
Technology Officer, Audience
In recent years, advancements in noise suppression technology, and the
willingness to design a second microphone into a mobile handset have enabled
the last bastion of voice quality challenges heretofore unsolved —
background noise suppression.
With some of the basic connectivity and network coverage issues solved,
engineers are turning to new areas for audio quality improvement. With new
solutions, like Audience's, offering noise suppression of up to 30 dB, there is
a great opportunity in background noise suppression. With the addition of a
second microphone into the handset to capture audio signals, and a powerful
processor that emulates the human hearing system, mobile phone users are able
to hear and be heard in places previously unimagined.
When choosing an advanced noise suppression solution, there are several audio
quality enhancements that are now available. Transmit noise suppression in
handheld mode and speakerphone modes, acoustic echo cancellation, and receive
noise suppression with voice equalization that suppresses background noise from
the incoming caller, and dynamically boosts their voice over noises in the
local environment. Both objective and subjective measures are critical when
evaluating solutions. Objective measures are defined in the ITU-T G.160
specification, and subjective measures in the P.835 standard. The G.160
standard is being amended to accurately capture the high levels of suppression
that are being achieved in these new solutions. Improvements in Total Noise
Level Reduction (TNLR), Signal to Noise Ratio Improvement (SNRI), and Delta
Signal to Noise (DSN) are expected to be ratified by mid 2009.
The P.835 standard uses Mean Opinion Scores (MOS) ratings to measure audio
quality improvements of advanced noise suppressors. The standard has been newly
updated in Amendment I Appendix III calling for additional methods for testing
of non-stationary noise sources. |
| |
Improving Audio Performance in Handsets
By Sanjay Voleti, Audio Marketing Director, National
Semiconductor Corp.
Mobile handsets are packing in numerous new features such as video, MP3
playback, FM radio, GPS, and mobile TV driven by 3G networks. In addition, the
rapid growth of handset sales over the last decade means that these devices are
now used in a diverse range of environments (home, office, crowded airports,
train stations, etc.) with different ambient noise levels.
This all points to the need for today's handsets to support higher quality
audio while enabling phone conversations in all sorts of environments. Of
course, all this must be done with minimal impact, if any, on battery life.
There are several developments underway to address these trends.
For example, the latest audio subsystems now integrate ADCs, DACs as well as
headphone and speaker amplifiers. This level of integration does not sacrifice
performance — the ADCs and DACs support 24-bit audio and provide SNR
performance of 100 dB while the speaker amps are capable of providing output
power approaching 1.5W. Additionally, these audio subsystems provide very high
PSRR at the GSM pulse frequency to eliminate the "bumble bee" noise that can
interfere with surrounding audio equipment. Most of us have experienced this
noise when using older cell phones in a car or near other speaker systems.
Handset manufacturers also want users to enjoy extended battery life when
listening to MP3 music, FM radio or watching a video. New headphone amps based
on Class G or H architectures allow the headphone to use the most optimized
voltage supply, thereby helping extend battery life.
The most exciting developments are in the area of noise reduction. Everyone has
either seen or experienced a phone conversation affected by a very noisy
environment. Audio uplink and downlink noise reduction are directly focused on
addressing this issue. The key here is to make sure that natural voice quality
is preserved and power consumption is minimized.
Another issue handset makers wrestle with relates to the speakers. In order to
support usage in noisy environments, most manufacturers like audio subsystems
that provide very high output power. This is important for creating high SPL
across various types of handsets with different form factors — sliders,
bar-type, flip phone, etc. However, the speakers are very small and high output
power can damage them. Newer audio subsystems solve this problem with built-in
Automatic Level Control, enabling handsets with the highest audio SPL, and no
speaker damage or distortion.
The ability to provide high quality audio for both entertainment and voice will
become a key differentiator as handsets continue to support an increasing
number of applications. |
Quickfilter Technologies
©
2010
Advantage Business Media
|
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