參數資料
型號: TLV320AIC29IRGZ
廠商: Texas Instruments, Inc.
元件分類: Codec
英文描述: STWREO AUDIO CODEC WITH INTERGRATED HEADPHONE AND SPEAKER AMPLIFIERS
中文描述: STWREO音頻編解碼器集成耳機和音箱放大器
文件頁數: 26/85頁
文件大?。?/td> 1007K
代理商: TLV320AIC29IRGZ
SLAS494A DECEMBER 2005 REVISED JANUARY 2006
www.ti.com
26
Table 1. AGC Settings
MIC HEADSET INPUT
MIC HANDSET INPUT
CELL-PHONE INPUT
BIT
CONTROL
REGISTER
BIT
CONTROL
REGISTER
BIT
CONTROL
REGISTER
AGC enable
D0
01H
D0
1EH
D0
24H
Target gain
D7D5
01H
D7D5
1EH
D7D5
24H
Time constants (attack and decay time)
D4D1
01H
D4D1
1EH
D4D1
24H
Noise threshold
D13D11
24H
D13D11
24H
D13D11
24H
Noise threshold flag
D11
04H
D11
04H
D14
24H
Hysteresis
D10D9
1DH
D10D9
1DH
D10D9
24H
Debounce time (normal to silence mode)
D8D6
26H
D8D6
26H
D8D6
27H
Debounce time (silence to normal mode)
D5D3
26H
D5D3
26H
D5D3
27H
Max PGA applicable
D15D9
26H
D15D9
26H
D15D9
27H
Gain applied by AGC
D15D8
01H
D15D8
1EH
D14D8
1FH
Saturation flag
D0
04H
D0
04H
D7
1FH
Clip stepping disable
NOTE:All settings shown in Table 1 are located in Page 2 of control registers.
D3
06H
D3
06H
D8
24H
Stereo Audio DAC
Each channel of the stereo audio DAC consists of a digital audio processing block, a digital interpolation filter,
digital delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide enhanced
performance at low sample 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 x Fsref and
changing the oversampling ratio as the input sample rate is changed. For Fsref of 48 kHz, the digital deltasigma
modulator always operates at a rate of 6.144 MHz. This ensures that quantization noise generated within the
delta-sigma modulator stays within the frequency band below 20 kHz at all sample rates. Similarly, for Fsref
rate of 44.1 kHz, the digital delta-sigma modulator always operates at a rate of 5.6448 MHz.
Digital Audio Processing
The DAC channel consists of optional filters for de-emphasis and bass, treble, midrange level adjustment, or
speaker equalization. The de-emphasis function is only available for sample rates of 32 kHz, 44.1 kHz, and 48
kHz. The transfer function consists of a pole with time constant of 50ms and a zero with time constant of 15ms.
Frequency response plots are given in the
Audio Codec Filter Frequency Responses
section of this data sheet.
The DAC digital effects processing block consists of a fourth order digital IIR filter with programmable
coefficients (one set per channel). The filter is implemented as cascade of two biquad sections with frequency
response given by:
N0
2
N1
z
1
N2
z
2
32768
2
D1
z
1
D2
z
2
N3
2
N4
z
1
N5
z
2
32768
2
D4
z
1
D5
z
2
The N and D coefficients are fully programmable, and the entire filter can be enabled or bypassed. The
coefficients for this filter implement a variety of sound effects, with bass-boost or treble boost being the most
commonly used in portable audio applications. The default N and D coefficients in the part are given by:
N0 = N3 = 27619
N1 = N4 = 27034
N2 = N5 = 26461
D1 = D4 = 32131
D2 = D5 = 31506
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