OmniSurround, ST Dynamic Bass, SRS® WOW™,
SRS® TruSurround XT™ which is Virtual Dolby®
Surround and Virtual Dolby® Digital compliant
● Independent Volume / Balance for Loudspeakers
and Headphone
● Loudspeakers: Smart Volume Control (SVC),
5-band equalizer and loudness
● Headphone: Smart Volume Control (SVC), Bass-
Treble, Loudness and SRS® Tru B a s s™
■ Analog Audio Matrix
● 4 stereo inputs
● 3 stereo outputs
● THRU mode
■ Audio Delay for Audio Video Synchronization
● Embedded stereo delay up to 120 ms for lip-sync
function (up to 180 ms for tuner input)
● Independent delay on headphone and loudspeaker
channels
The STV82x7 family, based on audio digital signal
processors (DSP), performs high quality and advanced
dedicated digital audio processing.These devices
provide all of the necessary resources for automatic
detection and demodulation of analog audio
transmissions for European and Asian terrestrial TV
broadcasts.
STV82x7
Digital Audio Decoder/Processor
PRELIMINARY DATA
Virtual or true, multi-channel capabilities and easy digital
links make them ideal for digital audio low cost consumer
applications. Starting from enhanced stereo up to
independent control of 5 loudspeakers and a subwoofer
(5.1 channels), the STV82x7 family offers standard and
advanced features plus sound enhancements, spatial
and virtual effects to enhance television viewer comfort
and entertainment.
Typical Applications
● Analog and digital TV with virtual surround sound
● Analog and digital TV with multi-channel surround
1.1.1 Core Features ............................................................................................................................................8
1.1.2 Software Information .................................................................................................................................9
1.1.4 Electrical Features ...................................................................................................................................10
The STV82x7 is a multistandard TV sound demodulator and audio processor which integrates
SRS® WOW™, SRS® TruSurround XT™, Dolby® Pro Logic®, Dolby® Pro Logic II®,Virtual
Dolby® Surround (VDS) and Virtual Dolby® Digital (VDD) capability.
ST advanced algorithms such as ST OmniSurround, ST WideSurround, ST Dynamic Bass are
also available in this audio sound processor. ST OmniSurround is a certified Dolby® algorithm for
the Virtual Dolby® Digital (VDD) and the Virtual Dolby® Surround (VDS). When using VDD or
VDS, either a Dolby® Digital or a Pro Logic® (or Pro Logic II®) decoder is mandatory
respectively.
This chip performs automatic multistandard analog TV stereo sound identification and demodulation (no specific I²C programming is required). It offers various audio processing
functions such as equalization, loudness, beeper, volume, balance, and surround effects. It provides
a cost-effective solution for analog and digital TV designs.
The STV82x7 is perfectly suited to current and future digital TV platforms, based on audio/video
digital chips (STD2000, (DTV100 platform) and the future WorldWide iDTV one chip) which
include an internal digital decoder (MPEG, Dolby® Digital...). In the case where a Dolby® Digital
decoder is embedded in the audio/video digital chip, Virtual Dolby® Digital could be obtained.
For the CTV100/120 platform, the device is offered as an alternative solution to the first-generation
chassis that uses the STV82x6.
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STV82x7 General Description
®
Table 1: STV82x7 Version List
STV8247STV8257STV8267STV8277STV8287
Demodulation
AM/FM - Mono, FM 2-carrier
NICAM
Multi-Channel Capability
3 x I²S In or 1 I²S Out,
S/PDIF (Pass-thru)
5.1 Analog Out for Loudspeakers
Virtual Dolby® Surround
Virtual Dolby® Digital capability
Dolby® Pro Logic®
Dolby® Pro Logic II®
Audio Processing
SRS® WOW™
SRS® TruSurround XT™
ST Voice, ST Dynamic Bass,
ST WideSurround
ST OmniSurround
2
S
T
T
V
V
8
8
2
2
1
0
7
7
T
T
V
V
8
8
2
2
3
2
7
7
S
S
S
S
T
S
T
V
8
2
4
7
D
S
V
T
8
V
2
8
4
2
7
5
D
7
S
X
S
S
T
T
V
V
8
8
2
2
5
5
7
7
D
D
S
X
S
T
S
T
V
8
2
6
7
D
S
V
T
8
V
2
8
6
2
7
7
D
7
S
X
S
T
S
V
T
8
V
2
8
7
2
7
7
D
7
S
D
X
XXXXXXXXXXXXXXXX
XXXXXXXXXXXXXX
XXXXXXXX
XXXXXXX
XXXXXXXXXX
1
XXXXXXXX
XXXX
XX
XXXXX
XXXXXXXXXXXXXXXX
XXXXXXXXXXXX
S
T
S
V
T
8
V
2
8
8
2
7
8
D
7
S
D
X
XX
1. Virtual Dolby Digital capability is obtained with the use of external Dolby Digital decoder (for
example STD05x0).
2. When using VDD or VDS with ST OmniSurround or SRS TruSurround XTTM, either a Dolby®
Digital or a Pro Logic® (or Pro Logic II®) decoder is mandatory respectively.
Figure 1: Package Ordering Information
Order Code:
STV82x7 (Tray)
STV82x7/T (Tape & Reel)
0
8
P
F
Q
T
For Example: STV8257DSX/T will be delivered in Tape & Reel conditioning
7/149
General DescriptionSTV82x7
1.1STV82x7 Overview
1.1.1Core Features
● Single audio source processing:
— IF source and/or analog stereo input (SCART)
— one digital source with a maximum of 6 synchronous channels (5.1 is obtained across three
I²S)
● SIF input signal with Automatic Gain Control (AGC)
● Digital Demodulator with automatic standard detection and demodulation for AM, FM mono,
FM 2 carriers (German or Korean FM 2-carrier) and NICAM
● Audio processor working at 32 kHz, 44.1 kHz or 48 kHz with specific features:
— For Loudspeakers (L, R, L
Dolby® Pro Logic II ® Decoder with Bass Management
SRS® WOW™ or TruSurround XT™ including Virtual Dolby® Surround and Virtual Dolby®
Digital
ST WideSurround
ST OmniSurround
ST Dynamic Bass
5-band Equalizer or Bass-Treble
Loudness
Smart Volume Control
Volume/Balance/Soft-mute
Beeper
Video Processing Delay Compensation
— For Headphone:
SRS® Tr u B a s s ™
Smart Volume Control
Bass-Treble
Loudness
Volume/Balance/Soft-mute
Beeper
Video Processing Delay Compensation
● Shared outputs for headphone and loudspeakers (surround channels);
● Analog matrix with:
— five external inputs:
four SCART inputs (2 V
one analog mono input (0.5 V
— one internal input from a digital matrix via a DAC
— three external outputs (2 V
— one internal output for the digital matrix (using an internal ADC)
● Digital matrix with:
— three input modes (Demodulator/SCART, SCART only and I²S)
— three stereo outputs (Loudspeakers, Headphone and SCART)
● High-end audio DAC
● S/PDIF output for connection with an external amplifier/decoder
● Internal multiplexer for the S/PDIF output (to share the internal S/PDIF output and the S/PDIF
output generated by the external decoder of the digital broadcast)
, RS, SubW, C):
S
capable)
RMS
RMS
capable)
RMS
)
8/149
STV82x7 General Description
● Specific stand-by mode (Loop-through)
● Control by I²C bus (two I²C addresses)
● System PLL and Clock Generation using either a single quartz oscillator or a differential clock
input
1.1.2Software Information
The different software combinations are listed in Tab l e 2 .
Table 2: Input/Output Software Configurations
Output (Number of Channels)
Input (Number of Channels)
2 (+1)4 (+1)5 (+1)
1
2 (L and R)
2 (LT and RT)
4 (+1)
5 (+1)
ST WideSurround or
SRS® WOW™
ST WideSurround or
SRS® WOW™
ST WideSurround or
SRS® TruSurround XT™ or
ST OmniSurround or
Dolby® Pro Logic® + SRS®
TruSurround XT™ or
Dolby® Pro Logic® + ST
OmniSurround
SRS® TruSurround XT™ or
ST OmniSurround or
Downmix
SRS® TruSurround XT™or
ST OmniSurround or
Downmix
Dolby® Pro Logic®
No processing
DownmixNo processing
Note:In addition to the above sound processing, it is always possible to add ST Voice and also ST
Dynamic Bass algorithms.
Note:The SRS® TruSurround® and ST OmniSurround are approved by Dolby as Virtual Dolby Surround
(VDS) and Virtual Dolby Digital (VDD).
The SRS® TruSurround XT™ system is composed of:
● SRS® TruSurround®
● SRS® WOW™
The SRS® WOW™ system also includes:
● SRS® Dialog Clarity™
● SRS® TruBass™
1.1.3Device Input Modes
● Demodulator and SCART Mode (with output f
● SCART Only Mode (with output f
● I²S Mode (with output f
= 48 kHz)
S
= 32, 44.1 or 48 kHz)
S
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= 32 kHz)
S
General DescriptionSTV82x7
— External audio input interface using 3 x I²S (for decoded streams such as Dolby® Digital
and/or standard stereo streams)
1.1.4Electrical Features
Multi Power Supply: 1.8 V, 3.3 V and 8 V.
Power Consumption:
● lower than 1 W in Functional mode (full features)
● 200 mW in Loop-through mode corresponding to Switch-off of all digital blocks
1.2Typical Applications
The STV82x7 is specified to enable flexible, analog and digital TV chassis design (refer to Figure 2,
Figure 3, Figure 4 and Figure 5).
The main considerations are:
● all necessary connections between devices can be provided through the TV set,
● pseudo stand-by mode used to copy to VCR or the DVD sources when the TV set is OFF,
● possible application compatibility with STV82x6 (TQFP80 package) TV design,
● pin-to-pin compatibility with STV82x8 (TQFP80 package) TV design.
The STV82x7 is used to process a single audio source (analog or digital). However, it is possible to
process two audio sources simultaneously using an STV82x7 interconnection (two chips can be
easily connected).
In the case of a single audio source, it is possible to hear and record in the same time: the same
audio stream can be simultaneously output on headphone, loudspeakers, S/PDIF and the SCART
connectors.
Note:Headphone and loudspeakers can be used simultaneously for dual-language purposes or for
different sound settings (e.g. volume). In this case, certain restrictions occur (see Section 4.2: Audio
Processing).
For more connections, the SCART-to-SCART path can be used. The use of these full analog paths
implies that the sound is not digitally processed.
Note : components with * are only mandatory in case of Dolby certification
19/149
System ClockSTV82x7
2System Clock
The System Clock integrates 2 independent frequency synthesizers.
The first frequency synthesizer can be used in one of two modes:
● In Mode 1, it is used by the demodulator, and the frequecy is 49.152 MHz.
● In Mode 2, it is used by the I²S input and is synchronous with the input frequency
or 48 kHz) and the frequency is 49.152 MHz (for f
= 32 or 48 KHz) or 45.1584 MHz (for fS =
S
44.1 KHz).
The second frequency synthesizer is used by the DSP core and can be adjusted between 100 and
150 MHz depending on the application (around 106 MHz at reset value).
The default values are designed for a standard 27-MHz reference frequency provided by a stable
single crystal or an external differential clock signal (for example, from the STV35x0) depending on
the CLK_SEL pin configuration (CLK_SEL = 1 means a single crystal, 0 means an external
differential clock). The 27-MHz value is the recommended frequency for minimizing potential RF
interference in the application. The sinusoidal clock frequency, and any harmonic products, remain
outside the TV picture and sound IFs (PIF/SIF) and Band-I RF.
Note:A change in the reference frequency is compatible with other default I²C programming values,
including those of the built-in Automatic Standard Recognition System.
= 32, 44.1
(fS
20/149
STV82x7 Digital Demodulator
3Digital Demodulator
The Digital Demodulator (see Figure 10) is composed of two channels. The first channel
demodulates an FM or an AM signal. The second channel demodulates FM 2-carrier or NICAM
signals (stereo demodulation).
All channel parameters are programmed automatically by the built-in Automatic Standard Recognition System (Autostandard) in order to find the correct sound standard. Channels can also
be programmed manually via the I²C interface for very specific standards not included among the
known standards.
The Analog Sound Carrier IF is connected to the STV82x7 via the SIF pin. Before Analog-to-Digital
Conversion (ADC), an Automatic Gain Control (AGC) is performed to adjust the incoming IF signal
to the full scale of the ADC. A preliminary video rejection is recommended to optimize conversion
and demodulation performances. The AGC system provides a gain value allowing for a wide range
of SIF input levels and is activated for all standards, except L/L’. In this particular case, the sound
carrier is AM-modulated and an automatic level adjustment would only damage the transmitted
audio signal. A preset I²C parameter is provided to define the gain of the AGC used in Manual mode
(Registers AGC_CTRL and AGC_GAIN).
Note:For optimum AM demodulation performance, it is recommended to use the MONO Input.
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Digital DemodulatorSTV82x7
3.2Demodulation
The demodulation system operates by default in Automatic mode. In this mode, the STV82x7 is able
to identify anddemodulate any TV sound standard including NICAM and A2 systems (see
Ta bl e 4 ) without any external control via the I²C interface. It consists of the two demodulation
channels (Channel 1 = Mono Left and Channel 2 = Mono Right/Stereo) to simultaneously process
two sound carriers in order to handle all transmission modes (stereo and up to three mono
languages). The built-in Automatic Standard Recognition System (Autostandard) automatically
programs the appropriate bits in the I²C registers which are forced to Read-only mode for users (see
Section 12.1). The programming is optimized for each standard to be identified and demodulated.
Each mono and stereo standard can be removed (or added) from the List of Standards to be
recognized by programming registers AUTOSTD_STANDARD_DETECT and
AUTOSTD_STEREO_DETECT, respectively. The identified standard is displayed in register
AUTOSTD_STATUS and any change to standard is flagged to the host system via pin IRQ. This flag
must be reset by re-programming the MSBs of register AUTOSTD_CTRL while checking the
detected standard status by reading registers AUTOSTD_STATUS, NICAM_STAT and ZWT_STAT.
Moreover, the detection of Stereo mode during demodulation is also flagged in register
AUTOSTD_STATUS.
Important: L/L’ and D/K standards cannot be automatically processed because the same frequency
is used for the MONO carrier. An exclusive L/DK selection must programmed in register
AUTOSTD_CTRL. This may be externally controlled by detecting the RF modulation sign, which is
negative for all TV standards except L/L’.
To recover out-of standard FM deviations or the Sound Carrier Frequency Offset, additional I²C
controls are provided without interfering with the Automatic Standard Recognition System
(Autostandard).
DK-NICAM Overmodulation Recovery: Four different FM deviation ranges can be selected (via
register AUTOSTD_CTRL) for the DK standard while the Autostandard system remains active. The
maximum FM deviation is 500 kHz in DK Mono mode and 350 kHz in DK NICAM mode (limited by
overlapping FM and NICAM spectrum values). The demodulated signal peak level (proportional to
the FM deviation) is detected by the Peak Detector and written to registers PEAK_DET_L and
PEAK_DET_R. This value is used to implement Automatic Overmodulation Detection via an
external I²C control.
Important: Only the selection of the 50 kHz FM deviation standard is compatible with the other DKA2* standards (DK1, DK2 or DK3). These standards must be removed from the list of standards
(registers AUTOSTD_STANDARD_DETECT and AUTOSTD_STEREO_DETECT) when
programming larger FM deviations reserved only for DK-NICAM standards.
SystemSound Type
FM Mono5.5
B/G
FM/NICAM5.55.850275080J1740
Typ e
Name
Table 4: Recognized Standards
Carrier 1
(MHz)
Carrier 2
(MHz)
FM Deviation
Nom. Max. Over
De-
emphasis
Roll
-off
(%)
Pilot
Frequency
(kHz)
FM 2-Carrier A25.55.74227508050 µs54.6875
D/K
D/K1FM 2-Carrier A2*6.56.25850 µs54.6875
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FM Mono6.5
FM/NICAM6.55.850275080J1740
STV82x7 Digital Demodulator
Table 4: Recognized Standards (Continued)
SystemSound Type
D/K2FM 2-Carrier A2*6.56.74250 µs54.6875
D/K3FM 2-Carrier A2*6.55.74250 µs54.6875
FM Mono6.0
I
FM/NICAM6.06.552275080J17100
LAM/NICAM6.55.850J1740
FM Mono4.515275075 µs
M/N
FM 2-Carrier A2+4.54.72415275075 µs55.069
Typ e
Name
Carrier 1
(MHz)
Carrier 2
(MHz)
FM Deviation
Nom. Max. Over
De-
emphasis
Roll
-off
(%)
Pilot
Frequency
(kHz)
For Chinese TV transmissions (DK-NICAM) which are subject to overmodulation, different FM
deviations are proposed for sound demodulation.
Sound Carrier Frequency Offset Recovery: Both Mono and Stereo IF Carrier frequencies can be
adjusted independently (registers CAROFFSET1 and CAROFFSET2) within a large range (up to
120 kHz for standard mono FM deviations) while the Automatic Standard Recognition System
remains active. The frequency offset estimation is written in registers DC_REMOVAL_L and
DC_REMOVAL_R (Mono Left / Channel 1 and Mono Right / Channel 2, respectively) and can be
used to implement the Automatic Frequency Control (AFC) via an external I²C control.
Manual Mode: If required, the Automatic Standard Recognition System system can be disabled
(Manual mode) and the user can control all registers including those only controlled by the
Automatic Standard Recognition System function when active. Manual mode is selected in register
AUTOSTD_STANDARD_DETECT (bit LDK_SCK, I_SCK, BG_SCK and MN_SCK set to 0).
23/149
Dedicated Digital Signal Processor (DSP)STV82x7
4Dedicated Digital Signal Processor (DSP)
A dedicated Digital Signal Processor (DSP) takes charge of all audio processing features and the
low frequency signal processing features of the demodulator. The internal 24-bit architecture will
ensure a high quality signal treatment and an excellent dynamic.
4.1Back-end Processing
The “back-end” processing corresponds to the low frequency signal processing (32 kHz or higher
frequencies) of the demodulator and other inputs (I²S, ADC).
Figure 11 shows a flowchart of the back-end processing tasks. However, the figure shows that the
processing is only a SINGLE SOURCE PROCESSING flow (no processing is possible with
“Demod + SCART” and I²S inputs simultaneously) and that the selection of a headphone output
restricts the loudspeakers configuration to 2+1 instead of 5+1.
Figure 11: Back-end Audio Processing
“Demod + SCART” or “SCART only” Input Modes
FM Channel1
FM Channel2
NICAM L
NICAM R
SCART L
SCART R
“I2S” Input Mode
I2S in 1
I2S in 2
I2S in 3
Stereo Peak Detector: 9D, bit 7 = 1
SRC X2/X4
DC
Removal
DC
Removal
DC
Removal
Autostandard
FMFM
De-emphasis
NICAM
De-emphasis
Stereo Peak Detector: 9D, bit 7 = 1
FM
Prescale
NICAM
Prescale
SCART
Prescale
I²S
Prescale
Dematrix
NICAM
Dematrix
LS
2
(L and R)
HP
2
(L and R)
SCART
Stereo Peak Detector: 9D, bit 7 = 0
Digital Audio Matrix
(L and R)
2 to 6
(L,R,C,LFE,Ls,Rs)
(L and R)
2
LS
HP
2
DownMix
SCART
2
Stereo Peak Detector: 9D, bit 7 = 0
(L and R)
24/149
STV82x7 Dedicated Digital Signal Processor (DSP)
The main features depend on the path:
● FM Channel
— DC Removal
—Prescaling
— De-emphasis (50 or 75 us)
— Stereo Dematrix
● NICAM Channel
— DC Removal
—Prescaling
— De-emphasis (J17)
— Dematrix
● Input SCART Channel
— DC Removal
—Prescaling
● Input I²S Channel
— I²S Prescaling
● Digital Audio Matrix
— Audio Channel Multiplexer between the different sources (IF, I²S, SCART) towards all
outputs (S/PDIF, LS, HP or SCART).
● Autostandard management
— device configuration depending on the standard to be detected
— freeze the device when a standard is detected
— once a standard detected, check that there is no change in the detection status
— set the correct action depending on any change in the detection status (mono backup or
mute setup and new standard detection)
● SCART
— Downmixing: L
— Soft Mute
/ RT or L0 / R0 (see AC-3 specification)
T
4.2Audio Processing
The following software is provided for main loudspeakers (L, R, C, LS, RS, SubW):
● Downmix
● Dolby® Pro Logic II® Decoder (L
● ST WideSurround, ST OmniSurround, SRS® WOW™ or SRS® TruSurround XT® (certified
Virtual Dolby® Surround and Virtual Dolby® Digital)
● ST Dynamic Bass
● Smart Volume Control (SVC)
● 5-band Equalizer or Bass-Treble
● Loudness
● Volume with independent channels (Smooth Volume Control)
● Pink Noise Generator (used to position the loudspeakers)
● Programmable Delay for each loudspeaker
● Adjustable Delay for “lip sync” up to 120 ms (to compensate audio/video latency) in SCART
Only Mode and up to 180 ms in Demodulator and SCART Mode
The following software is provided for the headphone or auxiliary output:
● Downmix
● SRS® TruBass™
● Smart Volume Control (SVC)
● Bass/Treble
● Loudness
● Independent Volume for each channel (Smooth Volume Control)
● Soft Mute
● Balance
● Beeper
● Adjustable Delay for “lip sync” up to 120 ms (to compensate audio/video latency) in SCART
Only Mode and up to 180 ms in Demodulator and SCART Mode
The following software is provided for SCART or S/PDIF outputs:
● Downmix
● Soft Mute
26/149
STV82x7 Dedicated Digital Signal Processor (DSP)
Figure 12: Audio Processing for Loudspeakers, Headphone, SCART and S/PDIF outputs
output Select
LS
S/PDIF
Output
SCART
Output
Output
Center
Output
Subwoofer
Output
Surround
Output
Headphone
Output
Input
S/PDIF
Digital
Vol um e
Bass/
SVC
SRS
S
Soft
Mute
Balance
ness
Loud-
Tr e bl e
Tr u B as s
Beeper
S
Digital
Soft
Vol um e
Bass /
Treble or
ST
Dynamic
SRS
Mute
Balance
5 bands
Bass
TruBass
Digital
Soft
Vol um e
Balance
Mute
Digital
Soft
Mute
Select
S/PDIF
Digital
Soft
Mute
Digital
Soft
Mute
Digital
Soft
Mute
2/0
Bass
Mgmt.
Loud-
and
3/2
ness
SVC
Vol um e
Vol um e
Balance
Vol um e
Equalizer
or
or
Pro Logic
C
ST
OmniSurnd
Decoder
Pro Logic II
LFE
Delay
Adjustable
L HP
Ls
Rs
R HP
SRS
XT
TruSurround
ST Wide
Surround
1to2/2to2
Dolby
Delay
L SCART
R SCART
Adjustable
L
R
27/149
Dedicated Digital Signal Processor (DSP)STV82x7
4.3ST WideSurround
STV82x7 offers three preset ST WideSurround Sound effects on the Loudspeakers path:
● Music, a concert hall effect
● Movie, for films on TV
● Simulated Stereo, which generates a pseudo-stereo effect from mono source
“ST WideSurround Sound” is an extension of the conventional stereo concept which improves the
spatial characteristics of the sound. This could be done simply by adding more speakers and coding
more channels into the source signal as is done in the cinema, but this approach is too costly for
normal home use. The ST WideSurround system exploits a method of phase shifting to achieve a
similar result using only two speakers. It restores spatiality by adding artificial phase differences.
The Surround/Pseudo-stereo mode is automatically selected by the Automatic Standard
Recognition System (Autostandard) depending on the detected stereo or mono source. By default,
“Movie” is selected for Surround mode. This value may be changed to “Music” by the
STSRND_MODE bit in the STSRND_CONTROL register.
Additional user controls are provided to better adapt the spatial effect to the source. The ST
WideSurround Gain (STSRND_LEVEL) and ST WideSurround Frequency (STSRND_FREQ)
registers can be used to enhance Music Predominancy in Music mode and Theater effect and Voice
Predominancy in Movie mode.
4.4ST OmniSurround
STV82x7 offers a spatial virtualizer to output any multi-channel input in stereo on the Loudspeakers
path:
“ST OmniSurround” will recreate a multi-channel spatial sound environment using only the Left and
Right front speakers. It can be adapted to any input configuration (OMNISRND_INPUT_MODE).
ST Voice will allow you to enhance the voice content of your program to increase the intellegibility
and the presence of the sound.
4.5Dolby Pro Logic II Decoder
Dolby® Pro Logic II® is a matrix decoder that decodes the five channels of surround sound that
have been encoded onto the stereo sound tracks of Dolby® Surround program material such as
DVD movies and TV shows.
It is even possible to decode standard stereo signals like music or non encoded movies.
Furthermore, it is an active process designed to enhance sound localization through the use of very
high-separation decoding techniques.
The Dolby® Pro Logic II® decoder is also able to emulate the former Dolby® Pro Logic® decoder
in a specific mode.
4.6Bass Management
This processing will generate the subwoofer signal and adjust all loudspeakers channels gain and
bandwidth.
Speakers capable of reproducing the entire frequency range will be referred to as “full range
speakers”, then signals sent to full range speaker will be full bandwidth (no filtering).
28/149
STV82x7 Dedicated Digital Signal Processor (DSP)
Speakers that have limited bass handling capabilities will be referred to as “satellite speakers”, then
signals sent to satellite speaker will be high-pass filtered to remove bass information below 100 Hz.
In the STV82x7, five output configuration modes have been implemented according to “Dolby
Digital Consumer Decoder” specifications. They are described below.
4.6.1Bass Management Configuration 0
In some cases, the bass management filters are available in the decoder itself, so there is no need
to reproduce these filters. The output configuration shown in Figure 13 offers this possibility.
Figure 13: Bass Management Configuration 0 (with Pro Logic switch indicating its reset state)
L
R
C
Ls
Rs
LFE
-5 dB
-15 dB
+
L
R
C
Ls
Rs
SubW
29/149
Dedicated Digital Signal Processor (DSP)STV82x7
4.6.2Bass Management Configuration 1
Configuration 1, shown in Figure 14, assumes that all five speakers are not full range and that all of
the bass information will be redirected to and reproduced by a single subwoofer. This configuration
is intended for use with 5 satellite speakers.
To prevent signal overload, the five main channels are attenuated by 15 dB, while the LFE channel
is attenuated by 5dB to maintain the proper mixing ratio.
Figure 14: Bass Management Configuration 1 (with Pro Logic switch indicating its reset state)
L
R
C
Ls
Rs
LFE
-5 dB
-15 dB
+
L
R
C
Ls
Rs
SubW
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STV82x7 Dedicated Digital Signal Processor (DSP)
4.6.3Bass Management Configuration 2
Configuration 2 assumes that the left and right speakers, are full range while the center and
surround speakers are smaller speakers. Also, all bass data is redirected to the left and right
speakers.
This configuration include output level adjustment that allows 12 dB attenuation for the 3 smaller
speakers (C, Ls, Rs). When the level adjustment will be disabled the decoder boosts by 12 dB the
full range speakers (Left, Right).
The third configuration, shown in Figure 16, assumes that all speakers except the center are full
range, then all bass information will be directed to and reproduced by the front left and front right
and both surround speakers. In order to provide more flexibility to this configuration, a switch will
offer an option which will produce a subwoofer channel by the LFE channel.
When the Subwoofer Switch is OFF, the input channels will be attenuated by 8 dB. Configuration 3
is required in certain high-end products.
This configuration implements the Simplified Dolby configuration. The center, left surround and right
surround channels are summed and then filtered by the LPF. The composite bass information is
either summed back into the left and right channels or summed with the LFE channel and sent to
the subwoofer output, see Figure 17.
Figure 17: Implementation of the Bass Management Configuration 4 (Simplified Configuration)
Ls
Rs
LFE
L
C
R
-4.5dB
+
Subwoofer
ON Switch
-5dB
+
+
-10.5dB
+
L
C
R
Ls
Rs
SubW
4.7SRS WOW and TruSurround XT
The SRS® TruSurround XT™ is a processing system that can accept from 1 to 6 channels on input
and that will generate a 2-channel output signal.
This processing system includes the latest SRS® algorithms:
● SRS® WOW™
● SRS® TruSurround® (Multi-channel signal virtualizer)
4.7.1SRS TruSurround
The SRS® TruSurround® is a processing that can accept from 2 to 5 channels on input and that will
generate a 2-channel output signal.
SRS® TruSurround® uses Head-Related Transfer Function (HRTF) -based frequency tailoring of
(L/R) difference signals to extend the sound image out past the physical boundaries of the speaker
placements to surround channel information. These rear channel HRTF curves have much greater
peak to valley differences at center frequencies. These were chosen to cause rear channel
difference signals to virtualize farther behind the listener and directed to a different virtual position
as compared to front channel signals. Information that is equal (L+R) in the rear surround channels
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Dedicated Digital Signal Processor (DSP)STV82x7
is processed by an identical HRTF curve but mixed in at a much lower amount. This HRTF
processing of equal (L/R) signals was again used to virtualize information to the rear of the listener.
The SRS® TruSurround® is certified by Dolby Laboratories to be a Virtual Dolby® Digital and
Virtual Dolby® Surround.
4.7.2SRS WOW
The SRS® WOW™ is an a sound processing system including:
● SRS® 3D Mono/Stereo™
● SRS® Dialog Clarity™
● SRS® TruBass™
4.7.2.1 SRS 3D Mono/Stereo
This system is used to create a pseudo-stereo signal for mono inputs or a three-dimensional spatial
signal for stereo inputs.
4.7.2.2 SRS Dialog Clarity
This system is used to enhance dialog perception.
4.7.2.3 SRS TruBass
The SRS® Tr u Ba s s ™ audio enhancement technology provides deep, rich bass to small speaker
systems without the need for a subwoofer or additional extra physical components. For systems
with a subwoofer, TruBass™ complements and enhances bass performance. Psycho-acoustically,
when the human ear is presented with a low frequency sound signal that is missing the fundamental
harmonic, it will fill in the fundamental frequency based on the higher harmonics that are present.
By accentuating the second and higher frequency harmonics of the bass portion of a signal,
TruBass™ gives the perception of greatly improved bass response.
SRS® TruBass™ is implemented on loudspeakers path, headphone path or on both in parallel.
4.8Smart Volume Control (SVC)
The Smart Volume Control regulates the audio signal level before audio processing. This regulation
is necessary in order for the signal level to be independent from the source (terrestrial channels, I2S
or SCART), its modulation (AM, FM or NICAM) and annoying volume changes (advertising, etc.).
The Smart Volume Control works as an audio compressor/expander; i.e. when the input signal
exceeds the threshold level, a very rapid attenuation (-2 dB/ms) is applied to rescale the signal
down to the threshold value. When the input signal is below the threshold level, the previous
attenuation is reduced slowly in order to retrieve the original input level (0dB gain). If the input signal
is too low, an addition gain of 6 dB can be provided.
To personalize the action of the SVC, five parameters are available:
1. Threshold: Maximum quasi-peak level that can be expected on output
2. Peak measurement mode: Select the channel on which the peak measurement must be
performed (Left, Right, Center...)
3. Release time: Gain slope applied to the amplification phase
4. Expander switch: To allow a +6dB amplification of small signals in order to reduce the output
dynamic range
5. Make up gain: Allows compensation of the signal amplitude limitation thanks to a 0 to 24 dB
adjustable gain.
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STV82x7 Dedicated Digital Signal Processor (DSP)
The SVC is implemented on the loudspeakers path, headphone path or on both in parallel
(independent settings). Also, the SVC can be applied in six-channel mode (L, R, L
SubW).
, RS, C and
S
4.9ST Dynamic Bass
STV82x7 offers dynamic bass boost processing on the Loudspeakers path:
ST Dynamic Bass is a bass boost process that can dramatically increase the bass content of any
program without any output level saturation.
3 cutoff frequencies (BASS_FREQ) can be chosen, 100Hz, 150Hz and 200Hz to adapt the effect to
your loudspeakers. The amount of bass (BASS_LEVEL) can also be fine tuned in order to adapt the
effect loudness.
4.105-Band Audio Equalizer
The loudspeakers audio spectrum is split into 5 frequency bands and the gain of each of band can
be adjusted within a range from -12 dB to +12 dB in steps of 0.25 dB. The Audio Equalizer may be
used to pre-define frequency band enhancement features dedicated to various kinds of music or to
attenuate frequency resonances of loudspeakers or the listening environment. The Equalizer is
enabled by the LS_EQ_ON bit in the LS_EQ_BT_CTRL register. The gain value for Band X is
programmed in register EQ_BANDX_GAIN.
The 5-Band Audio Equalizer is exclusive with Bass-Treble control. Bit LS_EQ_BT_SW in register
LS_EQ_BT_CTRL is used to select either the 5-Band Audio Equalizer or the Bass-Treble control for
the Loudspeakers path.
Depending on the LS Equalizer or LS Bass-Treble value, the volume level can be clamped to the LS
output to prevent any possible signal clipping from occuring using the ANTICLIP_LS_VOL_CLAMP
bit in the VOLUME_MODES (D7h) register.
The gain of bass and treble frequency bands for Headphone can be also tuned within a range from
-12 dB to +12 dB in steps of 0.25 dB. It may be used to pre-define frequency band enhancement
features dedicated to various kinds of music. The Headphone Bass/Treble feature is enabled by
setting the HP_BT_ON bit in the HP_BT_CONTROL register. The Bass and Treble gain values are
adjusted in registers HP_BASS_GAIN and HP_TREBLE_GAIN, respectively.
Figure 18: Equalizer
Depending on the HP Bass-Treble value, the volume level can be clamped to the HP output to
prevent any possible signal clipping from occuring using the ANTICLIP_HP_VOL_CLAMP bit in the
VOLUME_MODES (D7h) register.
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Dedicated Digital Signal Processor (DSP)STV82x7
4.12Automatic Loudness Control
As the human ear does not hear the audio frequency range the same way depending on the power
of the audio source, the Loudness Control corrects this effect by sensing the volume level and then
boosting bass and treble frequencies proportionally to middle frequencies at lower volume.
While maintaining the amplitude of the 1 kHz components at an approximately constant value, the
gain values of lower and higher frequencies are automatically progressively amplified up to +18 dB
when the audio volume level decreases.The maximum treble amplification can be adjusted from
0 dB (first order loudness) to +18 dB (second order loudness) in steps of 0.125 dB. As the volume is
proportional to the external audio amplification power, the loudness amplification threshold is
programmable in order to tune the absolute level. The Loudspeakers Loudness function is enabled
by setting the LS_LOUD_ON bit in register LS_LOUDNESS. The Loudspeakers Loudness
Threshold and Maximum Treble Gain values are also programmed in this register. The Headphone
Loudness function is enabled by setting the HP_LOUD_ON bit in register HP_LOUDNESS. The
Headphone Loudness Threshold and Maximum Treble Gain values are also programmed in this
register.
The loudness cut-off frequency is 100 Hz.
4.13Volume/Balance Control
The STV82x7 provides a Volume/Balance Control for all output channels configuration (except for
S/PDIF) with different volume level per channel (L, R, C, L
(from +11.875 to -116 dB, in a dB linear scale with a 0.125 dB step) largely covers typical home
applications (approx. 60 dB) while maintaining a good S/N ratio.
, RS, SubW, SCART). Its wide range
S
Figure 19: Volume Control
+11.875 dB
-116 dB
Output Gain
Mute
00h3FFh
An extra Master Volume Control can apply an extra gain/attenuation on L, R, C, L
I²C Control
, RS and SubW
S
channels.
The Volume/Balance Control can operate in one of two different modes:
● In Differential mode (default value), the volume control is a common volume value for both the
Left and Right Loudspeakers or Headphone channels (see Figure 19) and complimentary
balance control is used (see Figure 20).
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STV82x7 Dedicated Digital Signal Processor (DSP)
● In Independent mode, the volume for the Left and Right channels for Loudspeakers or
Headphone is controlled independently.
Figure 20: Differential Balance
100%
Output Gain
Mute
4.14Soft Mute Control
The Digital Soft Mute is applied smoothly (20 ms for 120 dB range) to avoid any switch noise on
output. It is available on all output channels pairs:
● S/PDIF channel (Left/Right)
● SCART channels (Left/Right)
● Loudspeakers channels (Left/Right)
● Center
● Subwoofer
● Headphone/Surround channels (Left/Right)
Another soft mute (analog) is also available on each DAC output.
l
hanne
C
Right
200h1FFh
Left Channel
000h
I²C Control (10 bits)
4.15Beeper
The beeper is used to generate a tone on the Loudspeakers or/and Headphone outputs. The
beeper sound (square wave) is added to the audio signal which is attenuated by 20 dB. The beep
sound amplitude includes a smooth attack and decay to avoid any parasitic noise when starting and
stopping.
It can be used for various applications such as beep sounds for remote control, alarm clock or other
features.
The Beeper operates in one of two modes:
● Pulse mode (beep applications): A tone with a programmable short duration (0.1, 0.25, 0.5
and 1.0 s) is generated. Afterwards, the beeper is automatically disabled and the output is
switched back to the audio signal, see Figure 21.
● Continuous mode (alarm application): A tone with a programmable long duration is
generated. Its start and stop controls must be programmed by I²C, see Figure 22.
The Beeper function is enabled by setting the BEEPER_ON bit in register BEEPER_ON.
Beeper parameters are controlled in register BEEPER_MODE.
The beeper tone level and frequency are programmed in register BEEPER_FREQ_VOL. The level
(or volume) ranges between 0 dB and -93 dB in steps of 3 dB and the tone frequency ranges
between 62.2 Hz and 8 kHz in steps of 1 octave.
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Dedicated Digital Signal Processor (DSP)STV82x7
A beep generator is shared only by the Loudspeakers or Headphone outputs. Therefore, in the
event of simultaneous beeps when in Pulse mode, only the first beep will define the effective
duration that will be the same for both outputs.
Figure 21: Pulse Mode
BEEP_ON = 1
0.1, 0.25, 0.5 and 1.0 s
T predefined
62.5 Hz < f < 8 kHz
BEEP_ON = 1
62.5 Hz < F < 8 kHz
BEEP_ON = 0
Figure 22: Continuous Mode
BEEP_ON = 0
T defined by I²C write
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STV82x7 Analog Audio Matrix (In / Out)
5Analog Audio Matrix (In / Out)
The analog part of the audio matrix can be divided into two parts: the SCART input matrix and the
SCART output matrix.
Figure 23: SCART Input Matrix
S1in
S2in
S3in
S4in
MONO_in
2
Select
Audio ADC
The SCART input matrix is an input for the digital matrix (after the ADC) which select which source
will be sent to the DSP.
Figure 24: SCART1/2/3 Output Matrix
Digital
Matrix
S1in
S2in
S3in
S4in
Stereo DAC
MONO_in
Select or Mute
2
Soft
mute
S1out
The SCART output matrix selects the sound to output, which can be directly a SCART input or the
output of the DSP. A mute function is provided to switch off the outputs.
A soft-mute function is provided to avoid all spurious sounds when switching from one position to
another position.
The SCART 2 and 3 output matrices have the same functions as the SCART 1 output matrix.
The particularity of the matrix is to accept input signal of 2 V
and to have the capability to output
RMS
such level. In this case, the power supply must be 8 V.
The Mono audio input is able to accept signals with a 0.5 V
amplitude.
RMS
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I²S Interface (In / Out)STV82x7
6I²S Interface (In / Out)
The STV82x7 offers three input/output choices: one I²S input, three I²S inputs or one I²S output.
6.1I²S Inputs
The STV82x7 can interface with a digital sound decoder. In this case, the digital data can be input
at a speed of 0.384 Mbytes/s (3.072 MHz for a 48 kHz sampling frequency with 32 bits of data).In
compliance with Dolby® specifications, only the sampling frequency is subject to restrictions. All
other requirements are extracted from other various specifications.
Table 5: I²S Characteristics
Sampling Frequency (kHz)
Data Size16, 18*, 20*, 24*, 32
PCMCLK
1. means that the number is the number of effective bits but the transmission is with 32 bits.
2. 512 x fs is used by the DACs if 512 x fs is present.
8, 11.025, 12,16, 22.05, 24, 32, 44.1 and 48
S1
2
512 x f
The PCMCLK (possible clock for upsampling) is provided by the master which is the digital sound
decoder. A sample rate conversion (SRC) will be necessary in the second case (STV82x7 slave) in
order to have a fixed frequency output from this block (either 32 kHz, 44.1 kHz or 48 kHz).
Note:The SRC function is only available in single I²S input mode.
The I²S interface is used in two ways depending on the package:
1. The interface with one I²S (I²S_DATA0) connection (only stereo or stereo-coded Dolby® Pro
Logic®);
2. One interface with three I²S connections connected to the DSP to allow the processing of a
multi-channel signal (maximum of 6 channels).
Figure 25: I²S Block Diagram
I²S_DATA0
Input = 8 to 48 kHz
f
S
I²S_DATA1
f
Input = 32 to 48 kHz
S
I²S_DATA2
f
Input = 32 to 48 kHz
S
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SRC x 2
SRC x 4
Audio Processing
STV82x7 I²S Interface (In / Out)
Table 6: I²S Frequency Configuration
(Max. Number of Channels)
1 (I²S_DATA0)832.0x 4
1 (I²S_DATA0)1632.0x 2
1 (I²S_DATA0)11.02544.1x 4
1 (I²S_DATA0)22.0544.1x 2
1 (I²S_DATA0)1248.0x 4
1 (I²S_DATA0)2448.0x 2
Both standard and non-standard modes are available, see Figure 26.
6.2I²S Output
A digital stereo output (I²S compatible) is also available for routing the demodulated signal or a
converted input audio signal to an external device. In this case the I²S_DATA0 signal and all clock
signals are set as outputs by setting bit D6 in register RESET to 1. The STV82x7 I²S drives the
serial bus (SCLK, LR_CLK, I²S_DATA0) in master mode in 64.fs format with a sampling frequency
(f
) of 32 kHz. The I²S_PCM_CLK signal can be used as a master clock in 512.fs format if required
s
for the slave interface. Both standard and non-standard modes are available, see Figure 26.
I²S
fS Input (kHz)
fS Output (kHz)
after SRC
SRC Use
33232.0No
344.144.1No
34848.0No
Note:The Input and Output modes for I²S are exclusive.
Figure 26: I²S Data Format: Lch = LOW, Rch = HIGH (I²S Input or Output mode)
1/f
I²S_LR_CLK
I²S_SCLK
(= 64f
)
s
I²S_DATAx
(standard mode)
I²S_DATAx
(non-standard mode)
1
MSB
1
2
MSB
2
3
3
Lch
2422
23
LSB
2422
23
LSB
s
1
MSB
12
MSB
2
3
Rch
1
12
2
3
22
3
23 24
LSB
22
23 24
LSB
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S/PDIF Input/OutputSTV82x7
7S/PDIF Input/Output
An S/PDIF output is available for connection with an external decoder/amplifier. An internal
multiplexer allows selection of either the internal signal or the external signal connected on the
SPDIF input (for example, the signal provided by the external MPEG audio / Dolby Digital decoder).
The outputted internal signal can be selected from:
● L/R
● C/Sub
● HP or Surround
● SCART.
A mute facility is also provided on the SPDIF output.
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STV82x7 Power Supply Management
8Power Supply Management
A mixed supply voltage environment requires the following voltages:
● 3.3V capable inputs/outputs for digital pins;
● 1.8V digital core;
● 8V capable inputs/outputs for analog audio interfaces (capability to output 2 V
requirements);
● 3.3V for stereo ADC and DAC (analog part);
● 1.8V for stereo ADC and DAC (digital part);
● 1.8V for IF ADC and AGC.
These voltages will be delivered by the application with an accuracy of ±5%. For more information,
refer to Section 13.3: Power Supply Data.
Other specific DC voltages or features are provided:
● Voltage Reference and Biasing Generation (AGC, ADCs, DACs),
● Bandgap reference.
for SCART
RMS
8.1Standby Mode (Loop-through mode)
The STV82x7 provides a Loop-through mode configuration that bypasses IC functions via a SCART
I/O pin (Full Analog Path only). In this case, only a minimum power of 200 mW is required.
In Standby mode, the digital and analog power supplies are switched off, except for pins VCC_H,
VCC33_LS, VCC33_SC, and VCC_NISO which are used to maintain the SCART path with the last
configuration programmed by analog matrixing (register SCART1_2_OUTPUT_CTRL and
SCART3_OUTPUT_CTRL). When switching back to normal Full Power mode, all I²C registers are
reset except for those used in Standby mode to maintain the original configuration.
In Standby mode, the I²C bus does not operate. However, the bus can still be used by other ICs
since the I²C I/O pins (SDA and SCL) of the STV82x7 are forced into a high-impedance
configuration.
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Additional Controls and FlagSTV82x7
9Additional Controls and Flag
This logic contains:
● the headphone detection,
● the IRQ generation, signal to be output to the MCU,
● the I²C bus expander output pin.
9.1Headphone Detection
For headphone, the HP_DET input can be used to automatically mute the Loudspeakers and
Subwoofer outputs when the HP_LS_MUTE bit is set in register HEADPHONE_CONFIG (active
low). When a headphone is detected (the HP_DET
Each change on the HP_DET
9.2IRQ Generation
Four IRQs are generated by the STV82x7. On each IRQ generation, the IRQ pin is set to 1. The
pending IRQ status must be read at the I²2S address 81h and the acknowledge is done by writing 0
to this register.
pin generates an IRQ request to the microprocessor on the IRQ pin.
pin is set to 0) and the Mute function is enabled.
The four availables IRQs are:
IRQ0: The identified TV sound standard is displayed in register AUTOSTD_STATUS. Each change
in the detected standard is flagged to the host system via hardware pin IRQ. The flag must be reset
by re-programming the IRQ bit in register AUTOSTD_CTRL and then checking the detected
standard status by reading registers AUTOSTD_STATUS, NICAM_STAT, and ZWT_STAT.
IRQ1: This IRQ is enabled only in digital input mode. In case of I2S synchronisation loss, this IRQ is
set to 1.
IRQ2: This IRQ is set to 1 when the device detects any change on the HP Detection pin
(Headphone connection or deconnection).
IRQ3: On the STV82x7, same pins are used for both Headphone and Surround loudspeaker signal
output. A change in the Headphone configuration (HP active or not active) will lead to a signal
switch on those hardware pins. In order to ensure a smooth audio transition, the output is soft muted
before the signal is switched. The IRQ3 is then set to 1 to advise the master processor that the
signal has been switched and to request a HP/Srnd Ouput Un-Mute.
9.3I²C Bus Expander
Pin BUS_EXP can be used to control external switchable IF SAW filters or audio switches. This pin
can be directly programmed by register RESET.
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STV82x7 STV82x7 Reset
10STV82x7 Reset
All STV82x7 features are controlled via the I²C bus.
The STV82x7 can be "reset" in 2 ways:
1. By Software via the I²C bus: This clears all synchronous logic, except for the I²C bus registers.
2. By Hardware via the RESET pin: In addition to clearing all synchronous logic, the RESET input
(active on the low level) resets all the I²C bus registers to the default values listed below.
Loudspeaker/Headphone SVC OFF, 0 dB Reference Value
Loudspeaker SurroundOFF
Loudspeaker 5-Band Equalizer OFF, 0 dB (Flat Band)
Loudspeaker Loudness OFF
Headphone Bass/TrebleOFF, 0 dB (Flat Band)
Loudspeaker/Headphone Beeper-40 dB / OFF
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I²C InterfaceSTV82x7
11I²C Interface
11.1I²C Address and Protocol
The STV82x7 I²C interface works in Slave mode and is fully compliant with I²C standards in Fast
mode (maximum frequency of 400 kHz). Two pairs of I²C chip addresses are used to connect two
STV82x7 chips to the same I²C serial bus. The device address pairs are defined by the polarity of
the ADR_SEL pin and are listed in the following table:
Table 8: I²C Read/Write Addresses
ADRWrite Address (W)Read Address (R)
LOW (connected to GND1)80h81h
HIGH (connected to VDD1)84h85h
Protocol Description
● Write Protocol
StartW ASub-addressAData A....A DataA Stop
● Read Protocol
StartW ASub-addressAStopStartR A DataA....ADataN
● W = Write address,
● R = Read address,
● A = Acknowledge,
● N=No acknowledge.
● Sub-address is the register address pointer; this value auto-increments for both write and read.
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STV82x7 I²C Interface
11.2Start-up and Configuration Change Procedure
Figure 27: Flow chart
Powe r ON
Hardware Reset (by pin 43)
Clock PLLs progammation
(for Crystal value different than 27 MHz)
Load Patch File
HW_RESET bit = 1
(bit 2 in HOST_CMD register)
INIT_MEM bit ?
(bit 0 in DSP_STATUS
register)
=1
Device Configuration Set-up
NOTE: This HW reset after Power ON is
mandatory to avoid bad device configuration
(FS1 & FS2 registers)
(by I²C transfer)
(DSP RUN)
(DSP inititialization)
=0
(analog or digital)
HOST_RUN bit = 1
(bit 0 in DSP_RUN register)
INIT_MEM bit = 0
HOST_NO_INIT bit = 1
(bit 1 in DSP_RUN register)
(OPTIONAL)
HOST_RUN bit = 0
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(start DSP processing)
(change configuration)
(registers 85h to FFh are not reset)
(stop DSP processing)
Register ListSTV82x7
12Register List
Note:The unused bits (defined as ‘Reserved’) in the I²C registers must be kept to zero.
The system clock registers (from address 08h to 0Bh and from address 5Ah to 5Dh) do not need to
be modified if a standard 27 MHz quartz crystal oscillator is used.
The default values of the demodulator registers (from address 0Ch to 55h) are for optimum
performances and any change is not recommended, except for:
● AGC_GAIN (0Fh) to adjust AGC gain for AM carrier in L/L' standard (AGC used in open loop).
● CAROFFSET1 (22h) and CAROFFSET2 (3Ah) to compensate IF carrier frequency with an
(96h) and PRESCALE_SCART (97h) to equalize demodulated or external audio signal before
audio processing. Peak detector registers PEAK_DET_INPUT (9Dh), PEAK_DET_L (9Eh),
PEAK_DET_R (9Fh), PEAK_DET_L_R (A0h) can be used to measure internal sound level.
Sound source selection for each audio output channel Loudspeakers, Headphone and SCART to
be done using AUDIO_MATRIX_INPUT (A2h).
In Multi-lingual mode, AUDIO_MATRIX_LANGUAGE (A4h) selects separately the language for
each audio output channel.
Register AUTOSTD_CTRL (8Ah) is used to select between L/L' or D/K/K1/K2/K3 standard which
can be discriminated automatically. To be used also to change maximum FM deviation (125 kHz, by
default) in case of wide overmodulation. AUTOSTD_STANDARD_DETECT (8Bh) and
AUTOSTD_STEREO_DETECT (8Ch) to define the list of mono and stereo standards to be
recognized automatically.
Note:() used in reset value column means that the bit or the byte is read-only.
(S) symbol indicates that the field value is represented in signed binary format.
(*) The field AGC_ERR[4:0] (AGC_GAIN) can be written by user if the bit AGC_CMD (AGC_CTRL)
is set to one (by default controlled by Automatic Standard Recognition System). To be used to
adjust manually the input gain of analog AGC amplifier for AM carrier (L/L').
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STV82x7 Register List
12.1I²C Register Map
By default, all I²C registers controlled by Automatic Standard Recognition System (Autostandard)
are forced to Read-only mode for the user. These registers and bits are shaded in Ta bl e 9 .
The built-in Automatic Standard Recognition System (Autostandard) can be disabled. In this case,
the Software Reset function (bits SOFT_LRST1 and SOFT_LRST2) can be used to implement the
Automatic Standard Recognition by I²C Software. This is not required if the built-in Automatic
Standard Recognition System function is used (default).
Bit NameResetFunction
BUS_EXP0Static control by I2C of hardware pin BUS_EXP
I²S_OUTPUT00 = I²S Input (I²S output will be provided on I2S_DATA0 pin)
Bit[5]0Reserved.
EN_STBY0Standby mode enabling
Bit 30Reserved.
SOFT_LRST20Softreset (active high) of Channel 2 detectors only.
1 = I²S Output (512 x fs will be provided on I2S_PCM_CLK pin)
0: Normal mode
1: To lock the digital signals before to settle the device in standby mode
SOFT_LRST10Softreset (active high) of Channel 1 detectors only.
SOFTR_RST0General softreset (active high) to reset all hardware registers except for I²C data.
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Register ListSTV82x7
I2S_CTRLI2S Synchronization Control Register
Address: 04h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000000LR_OFFLOCK_FLAG
Bit NameResetFunction
Bits[7:2]0Reserved.
LR_OFF0LR Signal Detection
0: LR signal detected and correct
1: Missing LR pulses detected
LOCK_FLAG0Lock Flag allowing unmute of Audio Output
I2S_STATI²S Synchronization Status Register
Address: 05h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000000LR_OFFLOCK_FLAG
Bit NameResetFunction
Bits[7:2]0Reserved.
LR_OFF0LR Signal Detection
0: LR signal detected and correct
1: Missing LR pulses detected
LOCK_FLAG0Lock Flag allowing unmute of Audio Output
I2S_SYNC_OFFSETI²S Synchronization Offset Frequency Register
Address: 06h
Type: R/W
12.3Clocking 1
A low-jitter PLL Clock is integrated and can be fully reprogrammed using the registers described
below. By default, the programming is defined for a 27-MHz quartz crystal frequency, which is the
frequency recommended for reducing potential RF interference in the application. However, if
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STV82x7 Register List
necessary, the PLL Clock can be re-programmed for other quartz crystal frequencies within a range
from 23 to 30 MHz. Other quartz crystal frequencies can be programmed on your demand.
Note:A Crystal Frequency change is compatible with other default I²C programming including the built-in
0 : SIF & SCART input (32 kHz)
1 : SCART input only (48 kHz)
2 : I2S input only
FS1_DIVFS1 I/O Divider Programming Register
Address: 08h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
EN_PROG0NDIV1[1:0]0SDIV1[2:0]
Bit NameResetFunction
EN_PROG0FS1 programmation enable
0: FS1 I2C registers programmation ignored by system - FS1 pre-programmed automatically by
SYS-CONFIG register (normal use with standard quartz of 27 MHz)
1: FS1 I2C registers programmation used by system - FS1 pre-programmation by SYS-CONFIG
desactivated (to be used in case of no standard quartz, different from 27 MHz)
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Register ListSTV82x7
Bit NameResetFunction
Bit 60Reserved.
NDIV1[1:0]01FS1 Input clock divider selection
Bit 30Reserved.
SDIV1[2:0]010FS1 Output clock divider selection
FS1_MDFS1 Coarse Selection Register
Address: 09h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000MD1[4:0]
Bit NameResetFunction
Bits[7:5]000Reserved.
MD1[4:0]10001 FS1 Coarse Selection
FS1_PE_HFS1 Fine Selection Register (MSBs)
Address: 0Ah
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
PE_H1[7:0]
Bit NameResetFunction
PE_H1[7:0]0011
FS1 Fine Selection (MSBs)
0110
FS1_PE_LFS1 Fine Selection Register (LSBs)
Address: 0Bh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
PE_L1[7:0]
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STV82x7 Register List
Bit NameResetFunction
PE_L1[7:0]0000
FS1 Fine Selection (LSBs)
0000
12.4Demodulator
DEMOD_CTRLDemodulator Control Register
Address: 0Ch
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
00FAR_MODE
Bit NameResetFunction
bit [7:6]000Reserved
FAR_MODE01: Farrow and Mono filter for NICAM active
GAP_MODE0Defines the clock gapping mode of the demodulator
0: (default), the FS1 freq is controlled by stl-error (clock-pll mode) to align the instantaneous
value of the internal clock with respect to the received NICAM clock
1: the FS1 freq is fixed and the mean value of the internal clock is aligned by variable gapping
(src-error) with respect to the received NICAM clock
GAP_MODEAM_SELDEMOD_MODE[2:0]
AM_SEL0Demodulator Configuration Select
0: FM configuration of demodulator (Default)
1: AM configuration of demodulator
DEMOD_MODE[2:0]110Demodulator Mode Select
CH1 FM
000:NormalFM Normal
001:WideFM Wide
010:NormalQPSK System B/G/L/D/K
011:WideQPSK System B/G/L/D/K
100:NormalFM Wide
101:WideFM Normal
110:NormalQPSK System I
111:WideQPSK System I
CH2 FM/QPSK
DEMOD_STAT Demodulator Detection Status Register
Address: 0Dh
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000QPSK_LKFM2_CARFM2_SQFM1_CARFM1_SQ
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Register ListSTV82x7
Bit NameResetFunction
Bit [7:5]000Reserved.
QPSK_LK0QPSK Lock Detection Flag
0: Not detected
1: Detected
FM2_CAR0Channel 2 FM/AM Carrier Detection Flag
0: Not detected
1: Detected
FM2_SQ0Channel 2 FM Squelch Detection Flag
0: Not detected
1: Detected
FM1_CAR0Channel 1 FM/AM Carrier Detection Flag
0: Not detected
1: Detected
FM1_SQ0Channel 1 FM Squelch Detection Flag
0: Not detected
1: Detected
Note:These registers allow direct access to the demodulator signal detectors.
AGC_CTRLIF AGC Control Register
Address: 0Eh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
AGC_CMD00AGC_REF[2:0]AGC_CST[1:0]
Bit NameResetFunction
AGC_CMD0Automatic Gain Control Command Mode
Normally set to 0 enabling automatic mode. For L/L’ standards, the AGC should be switched off
due to the presence of the AM sound carrier. In this case, a fixed gain value should be set using
the AGCS register.
0: Automatic mode. AGC controlled by the Autostandard function. (Default)
1: Manual/Forced mode
Bits[6:5]00Reserved.
AGC_REF[2:0]100This bitfield is used to defines the clipping level which adjusts the allowable proportion of samples
at the input of the ADC which will be clipped. The AGC tries to maximize the use of the full scale
range of the ADC. The default setting gives a ratio of 1/256.
0: Normal signal
1: Signal too large and AGC is overloaded
SIG_UNDER0AGC Input SIgnal Lower Threshold
0: Normal signal
1: Signal too small and AGC is underloaded
When the AGC is in Automatic mode (AGC_CMD = 0), bits SIG_OVER and SIG_UNDER indicate
if the input signal is too small/large and the AGC is under/overloaded. This is useful when setting
the STV82x7 SIF input level.
Note:When AGC_CMD = 0, AGC_ERR[4:0] can be read -- indicating the input level. It can also be written
to -- presetting the AGC level which will then adjust itself to the final value.
When AGC_CMD = 1, the AGC is off and writing to AGC_ERR[4:0] directly controls the AGC
amplifier gain. Reading AGC_ERR just confirms the fixed value.
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Register ListSTV82x7
DC_ERR_IF DC Offset Status for IF ADC
Address: 10h
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
DC_ERR[7:0]
Bit NameResetFunction
DC_ERR[7:0]00000000 DC offset error of IF ADC output
12.5Demodulator Channel 1
CARFQ1H, CARFQ1M, CARFQ1LChannel 1 Carrier DCO Frequency
CRF1[7:0](00000000) Channel 1 Carrier Recovery Frequency
Displays the instantaneous frequency offset of the Channel 1 Baseband PLL Demodulator.
CETH1Channel 1 FM/AM Carrier Level Threshold
Address: 20h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
CETH1[7:0]
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STV82x7 Register List
Bit NameResetFunction
CETH1[7:0]00100000 This register is used to compare the carrier level in the channel and the threshold value. This
level is measured after the channel filter and is relative to the full scale reference level (0 dB).
This is used as part of the validation of an FM signal, if the carrier level is below the threshold,
the signal is considered to be non-valid.
SQTH1[7:0]00111100 The squelch detector measures the level of high frequency noise (> 40 kHz) and compares it to
the threshold level (SQTH). If the level is below this value, the S/N of the FM signal is
considered to be acceptable. Values are given for FM with standard deviation.
CAROFFSET1[7:0]00000000 This value is used to correct the carrier frequency offset of the incoming IF signal. Automatic
frequency control in FM mode can be implemented by registers DC_REMOVAL_L and
DC_REMOVAL_R.
A DCO frequency offset (in two’s complement format) is added to the pre-programming value
by AUTOTSD in the CARFQ1 registers (corresponding to the standard IF carrier frequency).
The programmable carrier offset ranges from -192 kHz to +190.5 kHz with a resolution of
1.5 kHz.
For standard FM deviation, the value displays by DC_REMOVAL_L and DC_REMOVAL_R can
be directly loaded in CAROFFSET1 to exactly compensate the carrier offset on Channel 1
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Register ListSTV82x7
12.6Demodulator Channel 2
IAGCRChannel 2 Internal AGC Reference for QPSK
Address: 25h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
IAGC_REF[7:0]
Bit NameResetFunction
IAGC_REF[7:0]10001000Sets the mean value of the internal AGC, used for QPSK demodulation. The default setting
corresponds to half full scale amplitude at the baseband PLL input.
IAGCCChannel 2 Internal AGC Time Constant for QPSK
0: Internal AGC is active
1: Internal AGC is disabled
These bits control the time per step (values given for QPSK mode). The default value defines the
optimum trade-off between fast settling time (for the fastest NICAM identification) and the noise
immunity (minimum BER degradation)
Displays the instantaneous frequency offset of the Channel 2 Baseband PLL
CETH2Channel 2 FM Carrier Level Threshold
Address: 38h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
CETH2[7:0]
Bit NameResetFunction
CETH2[7:0]00100000This register is used to compare the carrier level in the channel and the threshold value. This
level is measured after the channel filter and is relative to the full scale reference level (0 dB).
This is used as part of the validation of an FM signal, if the carrier level is below the threshold,
the signal is considered to be non-valid.
SQTH2[7:0] 00111100The squelch detector measures the level of high frequency noise (> 40 kHz) and compares it to
the threshold level (SQTH). If the level is below this value, the S/N of the FM signal is
considered to be acceptable. Values are given for FM with standard deviation.
CAROFFSET2[7:0] 00000000This value is used to correct the carrier frequency offset of the incoming IF signal. Automatic
frequency control in FM mode can be implemented by registers DC_REMOVAL_L and
DC_REMOVAL_R.
A DCO frequency offset (in two’s complement format) is added to the pre-programming value
by AUTOTSD in the CARFQ2 registers (corresponding to the standard IF carrier frequency).
The programmable carrier offset ranges from -192 kHz to +190.5 kHz with a resolution of
1.5 kHz.
For standard FM deviation, the value displayed by register DC_REMOVAL_R can be directly
loaded in register CAROFFSET2 to exactly compensate the carrier offset on Channel 2.
12.7NICAM Registers
NICAM_CTRLNICAM Decoder Control Register
Address: 3Dh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
00000DIF_POLECTMAE
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Register ListSTV82x7
Bit NameResetFunction
Bits[7:3]00000Reserved.
DIF_POL00: No polarity inversion (Default)
ECT0Error Counter Timer: Defines the NICAM error measurement period
MAE0Max. Allowed Errors. Defines the NICAM error decoding for mute function.
1: Polarity inversion of the differential decoding
0: 128 ms (Default)
1: 64 ms
0: 511 Max (Default)
1: 255 Max
NICAM_BERNICAM Bit Error Rate Register
Address: 3Eh
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
ERROR[7:0]
Bit NameResetFunction
ERROR[7:0]00000000NICAM Error Counter Value
NICAM_STATNICAM Detection Status Register
Address: 3Fh
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
NIC_DETF_MUTELOACBI[3:0]NIC_MUTE
Bit NameResetFunction
NIC_DET0NICAM Signal Detect
0: NICAM signal no detected
1: NICAM signal detected
F_MUTE0Frame Mute
0: No mute
1: Mute due to Superframe Alignment Loss
LOA0Loss of Frame Alignment Word (FAW)
0: No Alignment Lost
1: Frame Alignment Word Lost
CBI[3:0]0000Indicates the received NICAM control bits
NIC_MUTE0Indicates the NICAM decoder mute
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STV82x7 Register List
12.8Stereo Mode
ZWT_CTRL Zweiton Detector Control Register
Address: 40h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
LRST_TONE_OFF
STD_MODETHRESH[3:0]TSCTRL[1:0]
Bit NameResetFunction
LRST_TONE_OFF0Control of the reset of the tone detector
0: Periodical reset of tone detection enabled
1: Periodical reset of tone detection disabled
STD_MODE_C00: German standard (Default)
THRESH[3:0]1100Defines the threshold of the detector for pilot and tone frequencies.
TSCTRL[1:0]00Defines both the detection time and the error probability (reliability of the detection).
DIALOG_CLARITY0SRS Dialog Clarity algorithm is present when set.
TRUBASS0SRS Trubass algorithm is present when set.
TRUSURROUND0SRS Trusurround algorithm is present when set.
PRO_LOGIC0Dolby Pro Logic algorithm is present when set.
MULTICHANNEL0Multichannels output is present when set.
0: Dolby Pro Logic I
0
1: Dolby Pro Logic II
0: 1 I2S input
0
1: 3 I2S inputs
DSP_STATUSDSP Status Register
Address: 84h
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0000000INIT_MEM
Bit NameResetFunction
Bits[7:1]0000000 Reserved.
INIT_MEM
DSP Initialization
0
0: DSP is not initialized.
1: DSP is initialized.
DSP_RUNRegister Description
Address: 85h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
TEST_MODE00
Bit NameResetFunction
TEST_MODE_
INPUT[7:6]
00active in TEST_MODE = 1 (bypass processing)
0: I2S_0 copied to SCART and SPDIF outputs
1: I2S_1 copied to SCART and SPDIF outputs
2: I2S_2 copied to SCART and SPDIF outputs
HOST_
NO_INIT
HOST_RUN
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Register ListSTV82x7
Bit NameResetFunction
TEST_MODE[5:4]000: standard configuration
Bits[3:2]00Reserved
HOST_ NO_INIT00: I2C register table is initialized when we soft reset
HOST_RUN00: soft reset DSP
1: bypass processing configuration
2: Clock Loop test
1: I2C register table is not initialized when we soft reset
1: start DSP processing
I2S_IN_CONFIGI²S Configuration Register
Address: 86h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
LOCK_MODE
_EN
0SYNCLRCLK_START
Bit NameResetFunction
LOCK_MODE_EN
0: Disable Lock Mode for external I2S input
1
1: Enable Lock Mode for external I2S input
LRCLK_
POLARITY
SCLK_
POLARITY
DATA_CFGI2S_MODE
Bit 60Reserved.
I2S synchronisation:
SYNC0
LRCLK_START
LRCLK_POLARITY0Polarity of the left data
SCLK_POLARITY
DATA_CFG
I2S_MODE
0: Capture directly
1: Wait for synchro
according to LRCLK POLARITY, first data take:
0
0: Left
1: Right
0: Falling edge
1
1: Rising edge
0: LSB First
1
1: MSB First
0: Non standard mode
0
1: Standard mode (Refer to Figure 26)
AV_DELAYAudio/Video Delay Register
Address: 89h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
DELAY_TIMEDELAY_ON
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STV82x7 Register List
Bit NameResetFunction
DELAY_TIME
0000000
DELAY_ON0Audio/video delay is enabled when set.
Audio Delay Time
0000000: 0 ms
...
0111100: 60 ms (48kHz)
...
1011010: 90 ms (32kHz)
Note:AV_DELAY acts on both LS and HP paths simultaneously (same delay).
12.12 Automatic Standard Recognition
AUTOSTD_CTRLAutomatic Standard Recognition Control Register
Address: 8Ah
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000
FORCE_SQUE
LCH
SINGLE_SHOTDK_DEV[1:0]LDK_SW
Bit NameResetFunction
Bits[7:5]000Reserved.
FORCE_SQUELCH
SINGLE_SHOT
DK_DEV[1:0]
LDK_SW
Allow to force squelch detection
0
0: FM squelch is taken into consideration for MONO detection
1: FM squelch is not taken into consideration for MONO detection
Single Shot Mode Selection
0
0: Single Shot mode is not selected
1: Single Shot mode is selected
Selects FM deviation configuration to take into account of overmodulation in DK_NICAM standard.
00
00: FM 50 kHz (Default)10: FM 350 kHz
01: FM 200 kHz11: FM 500 kHz
Makes exclusive the auto search of DK/K1/K2/K3 and L/L’ standard
1
0: DK/K1/K2/K3 standard auto-search / L/L’ disabled
1: L/L’ standard auto-search / DK/K1/K2/K3 disabled
1
1. Single_Shot mode can be used before disabling the Automatic Standard Recognition
(Autostandard) to pre-program demodulator registers in a defined standard and reduce I²C
programming in Manual mode
Note:Only standard deviation FM 50K kHz is compatible with other D/K1/K2/K3 standards in Automatic
Standard Recognition Search mode.
FM deviation superior to 350 kHz will degrade strongly NICAM reception due to overlapping of FM
and QPSK IF spectrum in DK-NICAM standard.
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Register ListSTV82x7
L/L’ and DK/K1/K2/K3 standard cannot be discriminated in Automatic Standard Recognition Search
mode because the same frequency is used for the mono IF carrier.
AUTOSTD_STANDARD_DETECTAuto Standard Check Standard Register
Address: 8Bh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0
NICAM_C4_OFFNICAM_GAP_
MODE
NICAM_MON
O_IN
LDK_SCKI_SCKBG_SCKMN_SCK
Bit NameResetFunction
NICAM_C4_OFF
NICAM_GAP_MODE
NICAM_MONO_IN
LDK_SCK
I_SCK
BG_SCK
MN_SCK
0: Autostandard will consider the C4 bit for MONO backup
0
1: Autostandard will ignore the C4 bit for MONO backup
0: NICAM, fast search
1
1: NICAM, slow search (no perturbations on LEFT channel in search mode)
0: the MONO backup for NICAM comes from internal demodulator
0
1: the MONO backup for NICAM comes from MONO input
L/L’ or D/K Mono Standard Enable
1
0: Disabled
1: Enabled
I Mono Standard Enable
1
0: Disabled
1: Enabled
B/G Mono Standard Enable
1
0: Disabled
1: Enabled
M/N Mono Standard Enable
1
0: Disabled
1: Enabled
Note:Autostandard is off when all mono standards are disabled (LDK_SCK = 0, I_SCK = 0, BG_SCK = 0
and MN_SCK = 0).
AUTOSTD_STEREO_DETECT Auto Standard Check Stereo Register
Stereo Mode Detection flag activated when a stereo standard coming from the demodulator
selected on Loudspeakers output. Stereo transmission modes are:
0
- Zweiton Stereo Carrier AND Stereo Modulation (indifferently German or Korean standard)
- NICAM stereo with backup (CBI = 1000)
- NICAM stereo with no backup (CBI = 0000)
AUTO STD_ON
STEREO_SID[1:0]00
MONO_SID[1:0]00
STEREO_OK0STEREO STANDARD DETECTED
MONO_OK0MONO STANDARD DETECTED
Automatic Standard Recognition System Status
0
0: Automatic Standard Recognition System is OFF
1: Automatic Standard Recognition System is ON
Identification of the detected TV sound standard. See Tab l e 1 9 .
Table 19: TV Sound Standards
System
M/N4.5 (FM 27k)00XXX4.724 (Zweiton A2+)00
B/G5.5 (FM 50k)01
I6.0 (FM 50k)10XXX6.552 (NICAM 100%)00
Mono Sound
(MHz)
MONO_SID
[1:0]
LDK_SW
XXX5.85 (NICAM 40%)00
XXX5.742 (Zweiton A2)01
DK_DEV
[1:0]
Stereo Sound
(MHz)
STEREO_SID
[1:0]
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STV82x7 Register List
Table 19: TV Sound Standards
System
L6.5 (AM)
D/K
D/K1/K2/
K3
Mono Sound
(MHz)
6.5 (FM 50k)
6.5 (FM 200k)01
6.5 (FM 350k)10
6.5 (FM 500k)11
6.5 (FM 50k)
MONO_SID
[1:0]
11
LDK_SW
1XX5.85 (NICAM 40%)00
0
0XX5.85 (NICAM 40%)00
0XX6.258 (Zweiton A2*)01
0XX6.742 (Zweiton A2*)10
0XX5.742 (Zweiton A2*)11
DK_DEV
[1:0]
00
Note:X means don’t care.
12.13 Audio Preprocessing and Selection Registers
DC_REMOVAL_INPUTDC Removal Register
Stereo Sound
(MHz)
5.85 (NICAM 40%)00
STEREO_SID
[1:0]
Address: 90h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
00000DC_SCARTDC_NICAMDC_DEMOD
Bit NameResetFunction
Bits[7:3]00000 Reserved.
DC_SCART
DC_NICAM
DC_DEMOD
0: SCART input, DC removal inactive
1
1: SCART input, DC removal active
0: NICAM input, DC removal inactive
1
1: NICAM input, DC removal active
0: FM input, DC removal inactive
1
1: FM input, DC removal active
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Register ListSTV82x7
DC_REMOVAL_LFM DC Offset Left Registerl
Address: 91h
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
DC_REMOVAL_L[7:0]
Bit NameResetFunction
DC_REMOVAL_L[7:0]
0000 0000
Displays (in two’s complement format) the FM (or AM) DC offset level after demodulation on
channel 1 (and removed automatically).
In FM mode, the DC offset value gives a direct value of the carrier frequency offset which is
used to compensate the DCO with the CAROFFSET1 value in the event of an out-of-standard
offset. The range and the resolution depend upon the FM bandwidth programmed defined in
register BCOEFF1. See Ta bl e 2 0.
DC_REMOVAL_RFM DC Offset Right Register
Address: 92h
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
DC_REMOVAL_R[7:0]
Bit NameResetFunction
DC_REMOVAL_R[7:0]
0000 0000
Displays (in two’s complement format) the FM (or AM) DC offset level after demodulation on
channel 2 (and removed automatically).
In FM mode, the DC offset value gives a direct value of the carrier frequency offset which is
used to compensate the DCO with the CAROFFSET2 value in the event of an out-ofstandard offset. The range and the resolution depend upon the FM bandwidth programmed
defined in register BCOEFF2. See Ta bl e 2 0 .
Table 20: DC_REMOVAL_L/R Range and Resolution
FM modeRange (kHz)Resolution (kHz)
Small± 960.750
Standard & A2 Standard± 1921.5
Medium± 3843
Large± 7686
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STV82x7 Register List
PRESCALE_SELECTAM/FM Prescaling Select Register
Address: 93h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0000000
Bit NameResetFunction
Bits[7:1]0000000 Reserved.
AM_FM_SELECT00: FM prescale is applied to demodulator channels
1: AM prescale is applied to demodulator channels
PRESCALE_AMAM Prescaling Register
Address: 94h
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0PRESCALE_AM
Bit NameResetFunction
Bit 70Reserved.
AM_FM_
SELECT
PRESCALE_AM[6:0]0000000 -12 to + 24 dB AM prescaling to normalize the AM demodulated signal level before audio
processing. Auto level control can be implemented by I2C software using the Peak Level
Detector. (Default value = 0 dB)
0: Peak detector placed between FM/NICAM Dematrix and Audio Matrix or between I²S
Prescale and DownMix
1: Peak detector placed before DC removal (For input saturation detection)
0000 : 0 dBFS to -42 dBFS
0001 : -6 dBFS to -48 dBFS
0010 : -12 dBFS to -54 dBFS
0011 : -18 dBFS to -60 dBFS
...
00: AM/FM or I2S 010: SCART or I2S 2
01: NICAM or I2S 1
PEAK_DET_LPeak Level Detector Status Register (L channel)
Address: 9Eh
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
OVERLOAD_LPEAK_L[6:0]
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Register ListSTV82x7
Bit NameResetFunction
OVERLOAD_L[7]0Memorise overload on the peak detection. This field can be reset.
PEAK_L[6:0]00000000Displays the Absolute Peak Level of the audio source selected. The measured value is
updated continuously every 64 ms. The range varies linearly from the full scale (0 dB) down to 1/
256 of the full scale (-48 dB).
In AM/FM Mono mode, only the PEAK_L[7:0] value must be taken into account.
In FM Mono mode, the audio peak level range depends upon the programmed FM bandwidth.
The unique difference is that the measurement is done after Sound pre-processing (DC offset
removal, Prescaling, De-emphasis and Dematrixing).
In FM Stereo mode, the maximum value may be used to check if the incoming signal level is
correctly adjusted by the prescaling factor or if there are no FM overmodulation problems
(clipping).
Programmable values are listed in Ta bl e 2 0 .
PEAK_DET_RPeak Level Detector Status Register (R channel)
Address: 9Fh
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
OVERLOAD_RPEAK_R[6:0]
Bit NameResetFunction
OVERLOAD_R[7] 0Memorise overload on the peak detection. This field can be reset.
PEAK_R[7:0]0000000Displays the Absolute Peak Level of the audio source selected. The measured value is
updated continuously every 64 ms. The range varies linearly from the full scale (0 dB) down to
1/256 of the full scale (-48 dB).
For more information, refer to register PEAK_DET_L.
PEAK_DET_L_RPeak Level Detector Status Register (L - R)
Address: A0h
Type: R
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
OVERLOAD_L_RPEAK_L_R[6:0]
Bit NameResetFunction
OVERLOAD_L_R[7] 0Memorise overload on the peak detection. This field can be reset.
PEAK_L_R[7:0]0000000Displays the Difference between L and R (L - R) channels for the audio source selected.
For more information, refer to register PEAK_DET_L.
000: Pro Logic 1 Emulation (forced if DPL version)
001: Virtual (DPL2 version only)
010: Music (DPL2 version only)
PL2_MODES[2:0]000
PL2_ACTIVE0
011: Movie (standard) (DPL2 version only)
100: Matrix (DPL2 version only)
101: Custom (DPL2 version only)
110: not applicable (DPL2 version only)
111: not applicable (DPL2 version only)
0: Dolby Prologic 2 is not active
1: Dolby Prologic 2 is active
Table 26: Prologic II Decode Mode Configuration
PL2
Mode
0
1Virtual
2Music
Decode ModeDimensionCenter Width
Pro Logic
Emulation
301002
301010
xx0x11
98/149
Auto-
Balance
Panorama
Surround
Coherence
SUR
Filtering
STV82x7 Register List
Table 26: Prologic II Decode Mode Configuration (Continued)
PL2
Mode
3
Decode ModeDimensionCenter Width
Movie/
Standard
4Matrix
5Custom
301000
300011
xxxxxx
Auto-
Balance
Panorama
Surround
Coherence
Note:(x = user defined parameter)
PCM_SRND_DELAYRegister Description
Address: ABh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
000SNRD_DELAY[4:0]
Bit NameResetFunction
SUR
Filtering
Bits[7:5]000Reserved.
SNRD_DELAY[4:0] 00000Surround Channel Delay
range: 0 to 30 (in ms)
PCM_CENTER_DELAYRegister Description
Address: ACh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0000CENTER_DELAY[3:0]
Bit NameResetFunction
Bits[7:4]0000Reserved.
CENTER_DELAY[3:0] 0000
Center Channel Delay
range: 0 to 10 (in ms)
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Register ListSTV82x7
PRO_LOGIC2_CONFIGRegister Description
Address: ADh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
PL2_LFE00PL2_SRND_FILTER[1:0]
Bit NameResetFunction
Bits[7:6]00Reserved.
00: 0: Off
PL2_SRND_FILTR[1:0] 00
PL2_RS_POLARITY0
PL2_PANORAMA0
PL2_AUTOBALANCE0
01: 1: Shelf Filter (for music and matrix modes)
10: 2: 7kHz LP
11: 3: not applicable
0: Rs polarity normal
1: Rs polarity inverted
0: Panorama Off
1: Panorama On
0: Autobalance Off
1: Autobalance On
See Table 26: Prologic II Decode Mode Configuration for programmation of these bits depending on
the decode mode.
PRO_LOGIC2_DIMENSIONRegister Description
PL2_RS_
POLARITY
PL2_
PA NO R A MA
PL2_
AUTOBALANCE
Address: AEh
Type: R/W
Bit 7Bit 6Bit 5Bit 4Bit 3Bit 2Bit 1Bit 0
0PL2_C_WIDTH0PL2_DIMENSION
Bit NameResetFunction
Bit 70Reserved.
000: 0, no spread = OFF
001: 20
010: 28
PL2_C_WIDTH[2:0]000
Bit 30Reserved.
011: 36
100: 54
101: 62
110:69
111: 90, phantom
100/149
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