Texas Instruments TLV320AIC3104 Evaluation Module (EVM) and USB motherboard TLV320AIC3104EVM-K TLV320AIC3104EVM-K 데이터 시트
제품 코드
TLV320AIC3104EVM-K
SLAS510C – FEBRUARY 2007 – REVISED DECEMBER 2010
The following table lists several example cases of typical MCLK rates and how to program the PLL to achieve
f
f
S(ref)
= 44.1 kHz or 48 kHz.
f
S(ref)
= 44.1 kHz
MCLK (MHz)
P
R
J
D
ACHIEVED f
S
(ref)
% ERROR
2.8224
1
1
32
0
44,100
0
5.6448
1
1
16
0
44,100
0
12
1
1
7
5264
44,100
0
13
1
1
6
9474
44,099.71
–0.0007
16
1
1
5
6448
44,100
0
19.2
1
1
4
7040
44,100
0
19.68
1
1
4
5893
44,100.3
0.0007
48
4
1
7
5264
44,100
0
f
S(ref)
= 48 kHz
MCLK (MHz)
P
R
J
D
ACHIEVED f
S
(ref)
% ERROR
2.048
1
1
48
0
48,000
0
3.072
1
1
32
0
48,000
0
4.096
1
1
24
0
48,000
0
6.144
1
1
16
0
48,000
0
8.192
1
1
12
0
48,000
0
12
1
1
8
1920
48,000
0
13
1
1
7
5618
47,999.71
–0.0006
16
1
1
6
1440
48,000
0
19.2
1
1
5
1200
48,000
0
19.68
1
1
4
9951
47,999.79
–0.0004
48
4
1
8
1920
48,000
0
STEREO AUDIO ADC
The TLV320AIC3104 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 that an audio master clock be provided and appropriate audio clock generation be
set up within the device.
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 that an audio master clock be provided and appropriate audio clock generation be
set up within the device.
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.
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 f
initial sampling rate of 128 f
S
to the final output sampling rate of f
S
. The decimation filter provides a linear phase
output response with a group delay of 17/f
S
. The –3-dB bandwidth of the decimation filter extends to 0.45 f
S
and
scales with the sample rate (f
S
). The filter has minimum 75-dB attenuation over the stop band from 0.55 f
S
to 64
f
S
. Independent digital high-pass filters are also included with each ADC channel, with a corner frequency that
can be independently set.
Because of the oversampling nature of the audio ADC and the integrated digital decimation filtering,
requirements for analog antialiasing filtering are very relaxed. The TLV320AIC3104 integrates a second-order
analog antialiasing filter with 20-dB attenuation at 1 MHz. This filter, combined with the digital decimation filter,
provides sufficient antialiasing filtering without requiring additional external components.
requirements for analog antialiasing filtering are very relaxed. The TLV320AIC3104 integrates a second-order
analog antialiasing filter with 20-dB attenuation at 1 MHz. This filter, combined with the digital decimation filter,
provides sufficient antialiasing 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 page 0, registers 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 on
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 page 0, registers 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 on
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27
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