LISA-U series - System Integration Manual
UBX-13001118 - R17
Advance information
System description
Page 104 of 190
Reference
Description
Part Number – Manufacturer
R3
10 k
Ω
Resistor 0402 5% 0.1 W
RC0402JR-0710KL - Yageo Phycomp
R4, R5
2.2 k
Ω
Resistor 0402 5% 0.1 W
RC0402JR-072K2L – Yageo Phycomp
SPK
32
Speaker
Various manufacturers
U1
16-Bit Mono Audio Voice Codec
MAX - Maxim
Table 47: Example of components for audio voice codec application circuit
As in general, any external digital audio device compliant to the configuration of the digital audio interface of
the cellular module can be used. Various external audio codecs other than the one described in Figure 56 and
Table 47 can be used to provide voice capability. For example the Maxim MAX9867 audio codec, the Maxim
MAX9880A audio codec as well as many other parts produced by the same or other various manufacturers
(including Texas Instruments, Wolfson / Cirrus Logic, Freescale, etc) can be used instead of the Maxim MAX9860
audio codec described above. The application circuit must be implemented and configured appropriately
according to the specific device and application requirements.
Any external signal connected to the digital audio interface must be tri-stated or set low when the module
is in power-down mode and during the module power-on sequence (at least until the activation of the
V_INT
supply output of the module), to avoid latch-up of circuits and allow a proper boot of the module.
If the external signals connected to the cellular module cannot be tri-stated or set low, insert a multi
channel digital switch (e.g. Texas Instruments SN74CB3Q16244, TS5A3159, or TS5A63157) between the
two-circuit connections and set it to high impedance during module power down mode and during the
module power-on sequence.
If the I
2
S digital audio pins are not used, they can be left unconnected on the application board.
1.11.3
Voiceband processing system
The voiceband processing on the LISA-U modules is implemented in the DSP core inside the baseband chipset.
The analog audio front-end of the chipset is connected to the digital system through 16 bit ADC converters in
the uplink path, and through 16 bit DAC converters in the downlink path. External digital audio devices can be
interfaced directly to the DSP digital processing part via the I
2
S digital interface. The analog amplifiers are skipped
in this case.
Available audio signal processing algorithms are:
Speech encoding (uplink) and decoding (downlink).The following speech codecs are supported in firmware
on the DSP for speech encoding and decoding:
GERAN GMSK codecs
o
GSM HR (GSM Half Rate)
o
GSM FR (GSM Full Rate)
o
GSM EFR (GSM Enhanced Full Rate)
o
HR AMR (GSM Half Rate Adaptive Multi Rate - Narrow Band)
o
FR AMR (GSM Full Rate Adaptive Multi Rate - Narrow Band)
o
FR AMR-WB (GSM Full Rate Adaptive Multi Rate - Wide Band)
UTRAN codecs:
o
UMTS AMR2 (UMTS Adaptive Multi Rate version 2 – Narrow Band)
o
UMTS AMR-WB (UMTS Adaptive Multi Rate – Wide Band)
Mandatory sub-functions:
o
Discontinuous transmission, DTX (GSM 46.031, 46.041, 46.081 and 46.093 standards)
o
Voice activity detection, VAD (GSM 46.032, 46.042, 46.082 and 46.094 standards)
o
Background noise calculation (GSM 46.012, 46.022, 46.062 and 46.092 standards)
