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TSC2111
SLAS495A JUNE 2006 REVISED OCTOBER 2007
www.ti.com
27
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
These coefficients implement a shelving filter with 0 dB gain from dc to approximately 150 Hz, at which point
it rolls off to 3 dB attenuation for higher frequency signals, thus giving a 3-dB boost to signals below 150 Hz.
The N and D coefficients are represented by 16bit twos complement numbers with values ranging from –32768
to +32767. Frequency response plots are given in the Audio Codec Filter Frequency Responses section of this
data sheet.
Interpolation Filter
The interpolation filter upsamples the output of the digital audio processing block by the required oversampling
ratio. It provides a linear phase output with a group delay of 21/Fs.
In addition, the digital interpolation filter provides enhanced image filtering to reduce signal images caused by
the upsampling process that are below 20 kHz. For example, upsampling an 8-kHz signal produces signal
images at multiples of 8 kHz, i.e., 8 kHz, 16 kHz, 24 kHz, etc. The images at 8 kHz and 16 kHz are below 20
kHz and still audible to the listener, therefore, they must be filtered heavily to maintain a good quality output.
The interpolation filter is designed to maintain at least 65 dB rejection of images that land below 7.455 Fs. In
order to utilize the programmable interpolation capability, the Fsref should be programmed to a higher rate
(restricted to be in the range of 39 kHz to 53 kHz when the PLL is in use), and the actual FS is set using the
dividers in bits D5D3 of control register 00H/page 2. For example, if Fs = 8 kHz is required, then Fsref can be
set to 48 kHz, and the DAC Fs set to Fsref/6. This ensures that all images of the 8-kHz data are sufficiently
attenuated well beyond a 20-kHz audible frequency range. Passband ripple for all sample-rate cases (from 20
Hz to 0.45 Fs) is +0.06 dB maximum.