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STEREO AUDIO ADC
AUTOMATIC GAIN CONTROL (AGC)
TLV320AIC32
SLAS479A – AUGUST 2005 – REVISED SEPTEMBER 2005
OVERVIEW (continued)
Fsref = 44.1 kHz
MCLK (MHz)
P
R
J
D
ACHIEVED FSREF
% ERROR
13.0
1
7
5618
47999.71
0.0006
16.0
1
6
1440
48000.00
0.0000
19.2
1
5
1200
48000.00
0.0000
19.68
1
4
9951
47999.79
0.0004
48.0
4
1
8
1920
48000.00
0.0000
The TLV320AIC32 includes a stereo audio ADC, which uses a delta-sigma modulator with 128-times
oversampling in single-rate mode, followed by a digital decimation filter. The ADC supports sampling rates from 8
kHz to 48 kHz in single-rate mode, and up to 96 kHz in dual-rate mode. Whenever the ADC or DAC is in
operation, the device requires an audio master clock be provided and appropriate audio clock generation be
setup within the part.
In order to provide optimal system power dissipation, the stereo ADC can be powered one channel at a time, to
support the case where only mono record capability is required. In addition, both channels can be fully powered
or entirely powered down.
The integrated digital decimation filter removes high-frequency content and downsamples the audio data from an
initial sampling rate of 128 Fs to the final output sampling rate of Fs. The decimation filter provides a linear phase
output response with a group delay of 17/Fs. The –3 dB bandwidth of the decimation filter extends to 0.45 Fs
and scales with the sample rate (Fs). The filter has minimum 75dB attenuation over the stopband from 0.55 Fs to
64 Fs. Independent digital highpass filters are also included with each ADC channel, with a corner frequency that
can be independently set to three different settings or can be disabled entirely.
Because of the oversampling nature of the audio ADC and the integrated digital decimation filtering,
requirements for analog anti-aliasing filtering are very relaxed. The TLV320AIC32 integrates a second order
analog anti-aliasing filter with 20-dB attenuation at 1 MHz. This filter, combined with the digital decimation filter,
provides sufficient anti-aliasing filtering without requiring additional external components.
The ADC is preceded by a programmable gain amplifier (PGA), which allows analog gain control from 0 dB to
59.5 dB in steps of 0.5 dB. The PGA gain changes are implemented with an internal soft-stepping algorithm that
only changes the actual volume level by one 0.5-dB step every one or two ADC output samples, depending on
the register programming (see registers Page-0/Reg-19 and 22). This soft-stepping ensures that volume control
changes occur smoothly with no audible artifacts. On reset, the PGA gain defaults to a mute condition, and upon
power down, the PGA soft-steps the volume to mute before shutting down. A read-only flag is set whenever the
gain applied by PGA equals the desired value set by the register. The soft-stepping control can also be disabled
by programming a register bit. When soft stepping is enabled, the audio master clock must be applied to the part
after the ADC power down register is written to ensure the soft-stepping to mute has completed. When the ADC
powerdown flag is no longer set, the audio master clock can be shut down.
An automatic gain control (AGC) circuit is included with the ADC and can be used to maintain nominally constant
output signal amplitude when recording speech signals (it can be fully disabled if not desired). This circuitry
automatically adjusts the PGA gain as the input signal becomes overly loud or very weak, such as when a
person speaking into a microphone moves closer or farther from the microphone. The AGC algorithm has several
programmable settings, including target gain, attack and decay time constants, noise threshold, and maximum
PGA gain applicable that allow the algorithm to be fine tuned for any particular application. The algorithm uses
the absolute average of the signal (which is the average of the absolute value of the signal) as a measure of the
nominal amplitude of the output signal.
Note that completely independent AGC circuitry is included with each ADC channel with entirely independent
control over the algorithm from one channel to the next. This is attractive in cases where two microphones are
used in a system, but may have different placement in the end equipment and require different dynamic
performance for optimal system operation.
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