Datasheet

Decay Time
Input
Signal
Output
Signal
AGC
Gain
Attack
Time
STEREO AUDIO DAC
DIGITAL AUDIO PROCESSING
TLV320AIC32
SLAS479C – AUGUST 2005 – REVISED DECEMBER 2008 ............................................................................................................................................. www.ti.com
Figure 26. Typical Operation of the AGC Algorithm During Speech Recording
Note that the time constants here are correct when the ADC is not in double-rate audio mode. The time
constants are achieved using the Fsref value programmed in the control registers. However, if the Fsref is set in
the registers to, for example, 48 kHz, but the actual audio clock or PLL programming actually results in a different
Fsref in practice, then the time constants would not be correct.
The TLV320AIC32 includes a stereo audio DAC supporting sampling rates from 8 kHz to 96 kHz. Each channel
of the stereo audio DAC consists of a digital audio processing block, a digital interpolation filter, multi-bit digital
delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide enhanced
performance at low sampling rates through increased oversampling and image filtering, thereby keeping
quantization noise generated within the delta-sigma modulator and signal images strongly suppressed within the
audio band to beyond 20 kHz. This is realized by keeping the upsampled rate constant at 128 × Fsref and
changing the oversampling ratio as the input sample rate is changed. For an Fsref of 48 kHz, the digital
delta-sigma modulator always operates at a rate of 6.144 MHz. This ensures that quantization noise generated
within the delta-sigma modulator stays low within the frequency band below 20 kHz at all sample rates. Similarly,
for an Fsref rate of 44.1 kHz, the digital delta-sigma modulator always operates at a rate of 5.6448 MHz.
The following restrictions apply in the case when the PLL is powered down and double-rate audio mode is
enabled in the DAC.
Allowed Q values = 4, 8, 9, 12, 16
Q values where equivalent Fsref can be achieved by turning on PLL
Q = 5, 6, 7 (set P = 5 / 6 / 7 and K = 16.0 and PLL enabled)
Q = 10, 14 (set P = 5, 7 and K = 8.0 and PLL enabled)
The DAC channel consists of optional filters for de-emphasis and bass, treble, midrange level adjustment,
speaker equalization, and 3-D effects processing. The de-emphasis function is implemented by a programmable
digital filter block with fully programmable coefficients (see Page-1/Reg-21-26 for left channel, Page-1/Reg-47-52
for right channel). If de-emphasis is not required in a particular application, this programmable filter block can be
used for some other purpose. The de-emphasis filter transfer function is given by:
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