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參數(shù)資料
| 型號： | DMA2271 |
| 英文描述： | Consumer IC |
| 中文描述： | 消費性IC |
| 文件頁數(shù)： | 23/36頁 |
| 文件大?。?/td> | 496K |
| 代理商： | DMA2271 |
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DMA 2271, DMA 2280, DMA 2281
23
3.3.1. The Golay and PT Byte Decoder
The data format has changed now from data burst for-
mat (99 bits) to packet format (751 bits). The header of
each packet contains defined addresses for the different
sound and data services and four bits representing the
sound characteristics. The PT Byte of each packet dis-
tinguishes between sound and data packets. After cor-
rection of header and PT Byte with the Golay and PT
Byte decoder, this information is used for automatic con-
figuration of the DMA 2271, DMA 2280, and DMA 2281.
In addition, the Golay decoder is used for measuring the
bit error rate of the transmission channel. The bits in er-
ror in each packet header are accumulated over one
frame (82 packets). The sum is stored in IM bus register
206 (Table 4–2) and can be read by the CCU which may
control different muting functions.
3.3.2. The Address Comparator
The DMA 2271, DMA 2280, and DMA 2281 D2–MAC
Decoders are able to treat different sound services auto-
matically by decoding the address field of the packet
header. The two continuity bits CI1 and CI0 are used to
link successive packets of the same service in case of
a 120 Byte sound coding service.
Among the different coding characteristics all combina-
tions are possible. The user can select up to four sound
channels simultaneously by programming the sound
services via the IM bus (address 203, 194, 195 and 196).
These addresses are compared with the address of
each transmitted sound packet. At correspondence, this
packet is selected and decoded.
3.3.3. The Sound Decoder
The sound decoding section converts all types of se-
lected sound packets into a sequence of 14–bit sound
samples. The medium–quality channels are up–
sampled to the 32 kHz sampling frequency of the high–
quality channels, i.e. every sample of a medium–quality
channel is put out twice, the second time as a zero sam-
ple. The second part of the interpolation is performed in
the AMU 2481 Audio Mixer where two oversampling fil-
ters are provided. The error correction section uses a
range check and/or Hamming decoder, depending on
the sound coding mode. The Hamming decoder is able
to correct one error per sample and to detect double er-
rors. The range check uses the highly protected scale
factor bits to check the MSBs of each sample. Its error
correction and detection abilities are shown in Table
3–2.
Erroneous samples, i.e. samples with uncorrectable er-
rors, are concealed by replacement with interpolated ad-
jacent samples. The storage capacity for buffering the
sound samples during processing and for obtaining a
smooth, regular output of sound samples is provided by
an external 64–K DRAM. To ensure the continuity of out-
put sound samples in case of packet loss or packet gain,
the silence information is used and blocks of samples
corresponding to “silence” are repeated or omitted.
3.3.4. The Sound Multiplex
After extension from 14 bits to 16 bits, the sound sam-
ples of the four channels are loaded into a 64–bit shift
register and transferred to the AMU 2481 Audio Mixer
via a serial 3–lines S bus. Fig. 2–14 shows the S bus tim-
ing.
Table 3–2:
Error correction and detection
Scale
Factor
Protec-
tion
Range
Defective
Bits
Error
Correc-
tion
Error
Detec-
tion
linear:
111
1
–
–
–
110
2
X13, X12
–
1
101
3
X13, X12, X11
–
2
011
4
X13
→
X10
1
2
100
5
X13
→
X9
1
3
010
6
X13
→
X8
2
3
001
7
X13
→
X7
2
4
000
8
X13
→
X6
2
4
companded:
010
6
X9, X8
–
1
001
7
X9, X8, X7
–
2
000
7
X9, X8, X7
–
2
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