RHYTHM R3710
www.onsemi.com
8
SIGNAL PATH
There are two inputs into the audio signal path. The first
input is the front microphone and the second input can be a
second microphone or telecoil input as selected by a
programmable MUX. The front microphone input is
intended as the main microphone audio input.
Analog input signals should be ground referenced to
MGND (microphones, telecoils, DAI). MGND is internally
connected to GND to minimize noise, and should not be
connected to any external ground point.
The audio input is buffered, sampled and converted into
digital form using an A/D converter. The digital output is
converted into a selectable 32 kHz or 16 kHz, 20−bit digital
audio signal. Further IIR filter blocks process the
microphone signal. These are followed by four cascaded
biquad filters: pre1, pre2, pre3 and pre4. These filters can be
used for frequency response shaping before the signal goes
through channel and adaptive processing.
The channel and adaptive processing consists of the
following:
• Frequency band analysis
• 1, 2, 4, 6 or 8 channel WDRC
• 16 frequency shaping bands (spaced linearly at 500 Hz
intervals, except for first and last bands)
• 128 frequency band adaptive noise reduction
• Frequency band synthesis
After the processing the signal goes through two more
biquad filters, post1 and post2, which are followed by the
AGC−O block. The AGC−O block incorporates the
Wideband Gain and the Volume Control. There are also two
more biquad filters, post3 and post4, and the Peak Clipper.
The last stage in the signal path is the D/A H−bridge.
White noise can be shaped, attenuated and then added into
the signal path at two possible locations: before the Volume
Control (between the Wideband Gain and the Volume
Control) or after the Volume Control (between post 4 and the
Peak Clipper) as shown in Figure 1.
Functional Block Description
iSceneDetect 1.0 Environment Classification
The iSceneDetect feature, when enabled, will sense the
environment and automatically control the enhancement
algorithms without any user involvement. It will detect
speech in quiet, speech in noise, music, quiet and noise
environments and make the necessary adjustments to the
parameters in the audio path, such as ANR, WDRC and
FBC, in order to optimize the hearing aid settings for the
specific environment.
iSceneDetect will gradually make the adjustments so the
change in settings based on the environment is smooth and
virtually unnoticeable. This feature will enable the hearing
aid wearer to have an instrument which will work in any
environment with a single memory.
Evoke Advanced Acoustic Indicators
Advanced acoustic indicators provide alerting sounds that
are more complex, more pleasing and potentially more
meaningful to the end user than the simple tones used on
previous products. The feature is capable of providing
pulsed, multi−frequency pure tones with smooth on and off
transitions and also damped, multi−frequency tones that can
simulate musical notes or chords.
A unique indicator sound can be assigned to each of the
ten system events: memory select (A, B, C, or D), low
battery warning, digital VC movement and digital VC
minimum/maximum. Each sound can consist of a number of
either pure tones or damped tones but not both.
A pure tone sound can consist of up to four tones, each
with a separate frequency, amplitude, duration and start
time. Each frequency component is smoothly faded in and
out with a fade time of 64 ms. The start time indicates the
beginning of the fade in. The duration includes the initial
fade−in period. By manipulating the frequencies, start times,
durations and amplitudes various types of sounds can be
obtained (e.g., various signalling tones in the public
switched telephone network).
A damped tone sound can consist of up to six tones, each
with a separate frequency, amplitude, duration, start time
and decay time. Each frequency component starts with a
sudden onset and then decays according to the specified time
constant. This gives the audible impression of a chime or
ring. By manipulating the frequencies, start times,
durations, decays and amplitudes, various musical melodies
can be obtained.
Acoustic indication can be used without the need to
completely fade out the audio path. For example, the
low−battery indicator can be played out and the user can still
hear an attenuated version of the conversation.
Adaptive Feedback Canceller
The Adaptive Feedback Canceller (AFC) reduces
acoustic feedback by forming an estimate of the hearing aid
feedback signal and then subtracting this estimate from the
hearing aid input. The forward path of the hearing aid is not
affected. Unlike adaptive notch filter approaches, the AFC
algorithm does not reduce the hearing aid’s gain. The AFC
is based on a time−domain model of the feedback path.
The third−generation AFC (see Figure 5) allows for an
increase in the stable gain (see Note) of the hearing
instrument while minimizing artefacts for music and tonal
input signals. As with previous products, the feedback
canceller provides completely automatic operation.
NOTE: Added stable gain will vary based on hearing aid
style and acoustic setup. Please refer to the
Adaptive Feedback Cancellation Information
note for more details.
