參數(shù)資料
型號(hào)： PCD6001
廠商： NXP Semiconductors N.V.
英文描述： Digital telephone answering machine chip
中文描述： 數(shù)字電話應(yīng)答機(jī)芯片
文件頁數(shù)： 61/96頁
文件大小： 385K
代理商： PCD6001
2001 Apr 17
61
Philips Semiconductors
Product specification
Digital telephone answering machine chip
PCD6001
13 THE CODECs
13.1
Definitions
In the description of the CODECs, amplitude units in dB
are used. The following definitions apply:
dBm:
used for absolute analog signal power levels.
0 dBm equals 1 mW power dissipation in 600
. A
single sinewave signal with a power level of 0 dBm
corresponds to an RMS voltage value of 774.6 mV.
dBmp:
used for absolute analog signal power levels
with psophometric weighting according to “CCITT
Recommendation G.223” This unit is used to express
analog noise power levels.
dBm0:
used for relative digital signal power levels.
0 dBm0 is defined in “CCITT Recommendation G.711
(Section 4, Table 5)” It follows that the maximum digital
signal power level is 3.14 dBm0 (A-law). Thus
3.14 dBm0istheRMSvalueofasinewavesignalwhose
peaks just reach the full-scale of the digital code. For the
(internal) bitstream signal (output of ARS and DNS) the
positive full-scale value is a continuous stream of ‘ones’,
whereas the negative full-scale value is a continuous
stream of ‘zeroes’. For the (internal) digital 14 or 16-bit
words, represented in 2s complement (MSB first) the
positive full-scale value is a ‘zero’ followed by 13 or 15
‘ones’, whereas the negative full-scale value is a ‘one’
followed by 13 or 15 ‘zeroes’.
dBm0p:
used for relative digital signal power levels with
psophometric weighting according to “CCITT
Recommendation G.223”
dB:
is used for the signal level gain between any two
nodes within the speech path. As different signal
representations are used within the speech path, the
gain value depends on the used signal definitions.
dBp:
is used for the signal level gain between any two
nodes within the speech path with psophometric
weighting according to “CCITT Recommendation
G.223”
The uniform PCM reference point is the (virtual) signal
node in the DSP at the input of the PCM encoder for the
analog-to-digital speech path and the output of the PCM
decoder for the digital-to-analog speech path.
13.2
CODEC architecture
The PCD6001 is provided with two CODECs that perform
the analog-to-digital and digital-to-analog conversion of
speech signals. In Fig.29, the CODECs are the interface
between the external analog peripherals and the DSP.
CODEC1 is used for the line interface and CODEC2 is
used for the loudspeaker and the microphone.
The DTCON register bit DTCON.4 selects the input to
CODEC1 (LIFMIN1 or LIFMIN2).
The main CODEC functions are (refer to Fig.29):
AMP - Pre-amplifier
ARS - Analog Receive Sigma delta ADC
DDF - Digital Decimation Filter
DNS - Digital Noise Shaper
ATD - Analog Transmit DAC.
For CODEC1 the balanced line interface input is fed to the
ARS block that performs analog-to-digital conversion, the
gain of the input can be set to the amplification steps:
7, 23 and 35 dB (see Section 17.5 for typical/maximum
gain specifications). This programmable range is used by
the microcontroller on command of the DSP to perform
limit or automatic gain control. The analog data is
converted by ARS to a bit stream. The basic sampling
frequency (f
s
) is 8 kHz. The DDF decimates the bit stream
down to 16-bit linear PCM data. The DF has a gain of
3.14 dB (which has to be added to the programmable ARS
gain)to achieveauniform referencepointat theDSP input
for linear PCM data. Finally, the DSP will decimate this
data to 16-bit linear PCM data at a rate of 8 kHz.
The reverse operation is performed in the transmit path.
TheDSPproduces16-bitlinearPCMtotheDNS.TheATD
which is a DAC converts the bit stream into an analog
signal. The converter has a programmable amplification
range of 18 dB. This programmability is
12,
6, +0 and
+6 dB.
CODEC2 is built-up in a similar manner as CODEC1, the
only difference being the microphone amplifier before the
ADC. This will amplify the balanced analog (microphone)
signal in the receive path with a fixed +15 dB (see
Section 17.5 for exact gain specifications). For direct
connectivity of an external microphone, a software on/off
switchable supply voltage is available.
Several registers are available for the CODECS control:
DTCON: for selecting the input to CODEC1
(DTCON.4 = 0 means LIFMIN1 is selected,
DTCON.4 = 1 means LIFMIN2 is selected) and for
alternative gain settings (see Section 13.2.2)
CDVC1: the volume control register for CODEC1
CDVC2: the volume control register for CODEC2.
CDTRx: test mode control registers for both CODECS
PMTRx: test mode control registers for both CODECS
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