It is a digital audio sound processor used for thin TV. Digital signal processor is Rohm original DSP only for TV
sound signal processing, and it’s cost performance is excellent. Digital inputs are two lines. Output is digital output
corresponding to 2.1ch or play of sub-voice L/R signal.
●Features
■DSP Part
Data width: 32bit (Data RAM)
Quickest machine cycle:40.7ns (512fs,fs=48kHz)
Multiplier: 32 x 24 → 56bit
Adder: 32 + 32 → 32bit
Data RAM: 256 x 32bit
Coefficient RAM: 128 x 24bit
Sampling frequency: fs=48kHz
Master clock: 512fs
(24.576MHz,fs=48kHz)
■Input output I/F
2 stereo digital signal input port : 16/20/24bit (I2S,left-align,right-align)
2 stereo digital signal output port : 16/20/24bit (I2S,left-align,right-align), S/PDIF output
■Sound signal processing function for TV
Prescaler, DC cut HPF, channel mixer, P
TREBLE, pseudo stereo, surround, P
master volume, L/R balance, postscaler, output clipper, subwoofer output processing
2
(P
Volume,P2Bass,P2Treble are Rohm original sound effect functions.)
2 LRCKI I2S Audio LR signal input
3 SDATA1 I2S Audio data input 1
4 SDATA2 I2S Audio data input 2
5 RESETX Reset status with “L”
6 MUTEX_SP DAC mute signal input(*1)
7 MUTEX_DAC SP mute signal input(*1)
8 SCLI I2C Forwarding clock input
9 SDAI I2C Data input output
10 VSS1 Digital I/O GND
11 DVDDCORE Connect to REG15 terminal
12 REG15 Built-in regulator voltage output
13 LDOPOFF Built-in regulator POFF signal
input
14 ANATEST Analog test monitor terminal
15 VDD Digital I/O power supply
16 N.C
17 N.C
18 PLLFIL Filter connection terminal for
PLL
19 VSS2 Digital I/O GND
20 MODE Test mode selection input
N.C.:Non Connection
(*1):signal terminal is used with D class amplifier IC (BD5446EFV etc.) for input I2S made by Rohm.
(*2) :It connects with the lead frame of a package. Please use by OPEN or GND connection.
Type
-
D
D
D
B
B
B
F
E
-
G
G
G
-
-
G
A
Technical Note
No.
21ADDR I2C Slave address selection
22SPDIFO S/PDIF Signal output
23N.C
24N.C
25N.C
26N.C
27N.C
28SDAO 2 line serial data output(*1)
29SCLO 2 line serial clock output(*1)
30MUTEX_DACO DAC mute signal output(*1)
31MUTEX_SPO SP mute signal output(*1)
32RESETXO Reset signal output(*1)
33SDATAO2 I2S Audio data output 2
34SDATAO1 I2S Audio data output 1
35LRCKO I2S Audio LR signal output 1
36BCKO I2S Audio clock output 1
37SYSCLKO System clock output(*1)
38VSS3 Digital I/O GND
BU9409FV’s Digital Sound Processing(DSP) consists of special hardware most suitable to Thin TV.
BU9409FV uses this special DSP to perform the following processing.
Prescaler, DC Cut HPF, Channel Mixer, P
2
Volume(Perfect Pure Volume), BASS, MIDDLE, TREBLE,
Data width: 32 bit (DATA RAM)
Machine cycle:40.7ns (512fs, fs=48kHz)
Multiplier: 32×24 → 56 bit
Adder:32+32
Data RAM: 256×32 bit
→ 32 bit
MUX
Data
Input
RAM
MUX
0
M
U
X
Coefficient RAM:128×24 bit
Sampling frequency: fs=48kHz
Master clock:512fs (24.576MHz, fs=48kHz)
ADD
Acc
Digital signal from 16bit to 24bit is inputted to DSP,
Output
and it is extended by +8bit(+42dB) as overflow margin on the upper side.
The clip process is performed in DSP when the process exceeding this range is performed.
Input1
Input2
Pre
scaler
HPF
Channel
mixer
Digital Audio Processing Signal Flow
P2Volume
BASS
MIDDLE
TREBLE
Scaler
Pseude
stereo
1
surround
&
P2BassP2Treble
Channel
Channel
mixer
mixer
Scaler
LPF
7Band
P-EQ
2
3band
P-EQ
EVR
&
Blance
EVR
&
Balance
Post
scaler
&
Clipper
Post
scaler
&
Clipper
44--11.. PPrreessccaalleerr
When digital signal is inputted to audio DSP, if the level is full scale input and the process of surround or equalizer is
performed, then it overflows, therefore the input gain is adjusted by prescaler.
Adjustable range is +24dB to -103dB and can be set by the step of 0.5dB.
Prescaler does not incorporate the smooth transition function.
The DC offset component of digital signal inputted to the audio DSP is cut by this HPF.
The cutoff frequency fc of HPF is 1Hz, and first-order filter is used.
Default = 0
44--33.. CChhaannnneell mmiixxeerr
It performs the setting of mixing the sounds of left channel & right channel of digital signal inputted to the audio DSP.
Here the stereo signal is made to be monaural.
○Pulse sound detection and High-speed recovery function(functioning only at the time of transition of (2)<->(3))
2
Volume function makes the P2Volume also compatible with large pulse sounds (clapping of hands, fireworks & shooting
P
2
etc.) in addition to normal P
operation (R_RATE) is performed at 4 or 8 times the speed of normal attack operation or recovery operation.
Selection of using the pulse sound detection function.
Default = 0
Select Address Value Operational explanation
&h3BC[ 7 ]
Selection of operating times of Recovery Time (R_RATE) in the case of using the pulse sound detection function
Default = 0
Select Address Value Operational explanation
&h3C [ 3 ]
Selection of pulse sound detection time
Default = 0
Select Address Operational explanation
&h3C [ 6:4 ]
Setting of operating level of pulse sound detection function
Operation is started by the difference between the presently detected value and the last value as a standard.
Default = 0
Select Address Operational explanation
&h3C [ 2:0 ]
Example) Present detection level A : -10dB → 10^(-10/20) = 0.32 The last detection level B : -30dB → 10^(-30/20) = 0.032 A – B : 0.32 – 0.032 = 0.288 → Operating by the setting of command ”4” to ”7”.
Volume operation. When large pulse sound is inputted, attack operation (A_RATE) or recovery
0 Not using of pulse sound detection function
1 Using of pulse sound detection function
0
1
Operating at 4 times the speed corresponding to the setting time of
R_RATE
Operating at 8 times the speed corresponding to the setting time of
R_RATE
BASS of TONE Control can use Peaking filter or Low-shelf filter.
The setting is converted, in the IC, into digital filter’s coefficients (b0, b1, b2, a1, a2)by selecting the F
transmitted to coefficient RAM. The switching shock noise at the time of alteration of setting can be prevented by the smooth
transition function.
○BASS Control
Selection of filter types
Selection of smooth transition function
Selection of smooth transition time
Setting of smooth transition start
In the case of using the smooth transition function, after being transmitted, by the &h40[0] command, to the coefficient RAM
for smooth transition, the alteration of BASS’s coefficients is completed by using this command.
What is necessary is the time of waiting, which is more than the time selected by the setting of Bass smooth transition time,
from the time the BASS smooth transition start (&h4C[0] = “1”) is executed until the following command is sent. Please
make sure to perform the Bass smooth transition stop(&h4C[0] = “0”) after the smooth transition is completed.
MIDDLE of TONE Control uses Peaking filter.
The setting is converted, in the IC, into digital filter’s coefficients (b0, b1, b2, a1, a2)by selecting the F,Q and Gain, and
transmitted to coefficient RAM. The switching shock noise at the time of alteration of setting can be prevented by the smooth
transition function.
○MIDDLE Control
Selection of smooth transition function
Selection of smooth transition time
Setting of smooth transition start
In the case of using the smooth transition function, after being transmitted, by the &h44[0] command, to the coefficient RAM
for smooth transition, the alteration of MIDDLE’s coefficients is completed by using this command.
What is necessary is the time of waiting, which is more than the time selected by the setting of MIDDLE smooth transition
time, from the time the MIDDLE smooth transition start (&h4C[1] = “1”) is executed until the following command is sent.
Please make sure to perform the MIDDLE smooth transition stop(&h4C[1] = “0”) after the smooth transition is completed.
Setting of the Start of transmitting to coefficient RAM
In the case of using the smooth transition, it is transmitted to the coefficient RAM for smooth transition. In the case of not
using of the smooth transition, it is transmitted to the direct coefficient RAM.
TREBLE of TONE Control can use Peaking filter or High-shelf filter.
The setting is converted, in the IC, into digital filter’s coefficients (b0, b1, b2, a1, a2)by selecting the F
transmitted to coefficient RAM. The switching shock noise at the time of alteration of setting can be prevented by the smooth
transition function.
○TREBLE Control
Selection of filter types
Selection of smooth transition function
Selection of smooth transition time
Setting of smooth transition start
In the case of using the smooth transition function, after being transmitted, by the &h48[0] command, to the coefficient RAM
for smooth transition, the alteration of TREBLE’s coefficients is completed by using this command.
transition time, from the time the TREBLE smooth transition start (&h4C[2] = “1”) is executed until the following command is
sent. Please make sure to perform the TREBLE smooth transition stop(&h4C[2] = “0”) after the smooth transition is
completed.
,Q and Gain, and
0
Default = 0
Select Address Value Operational explanation
&h48 [ 7 ]
Default = 0
Select Address Value Operational explanation
&h48 [ 6 ]
Default = 0
Select Address Value Operational explanation
&h48 [ 5:4 ]
Default = 0
Select Address Value Operational explanation
&h4C [ 2 ]
What is necessary is the time of waiting, which is more than the time selected by the setting of TREBLE smooth
It is the deep bass equalizer making it possible that even thin-screen TV, by which the enclosure of speaker is restricted,
can reproduce the real sound close to powerful deep bass & original sound.
Solid & clear deep bass with little feeling of distortion is realized. Even boosting of bass does not interfere with vocal band,
therefore rich and natural deep band is realized.
Gain
ocal ban
P2Bass gain
P2Bassゲイン
ボーカル帯域
f
LPF Cutoff frequency
ON/OFF
2
Bass function
of P
Default = 0
HPF Cutoff frequency
HPFカットオフ周波数
LPFカットオフ周波数
Select Address Value Operational explanation
&h73 [ 7 ]
0 Not using of P2Bass function
1 Using of P2Bass function
2
Setting of P
Bass smooth transition time
Default = 0
Select Address Value Operational explanation
&h73 [ 3:2 ]
0 21.4ms
1 10.7ms
2 5.4ms
3 2.7ms
2
Bass smooth transition control
P
Default = 0
Select Address Value Operational explanation
&h77 [ 1:0 ]
0 P2Bass smooth transition stop
1 Setting of the values into Coefficient RAM for P2Bass smooth transition
2 P2Bass smooth transition start
2
What is necessary is the time of waiting, which is more than the time selected by the setting of P
2
time, from the time the P
Please make sure to perform the P
Bass smooth transition start(&h77[1:0] = “2”) is executed until the following command is sent.
2
Bass smooth transition stop(&h77[1:0] = “0”) after the smooth transition is completed.
Volume is from+24dB to -103dB, and can be selected by the step of 0.5dB.
At the time of switching of Volume, smooth transition is performed. The smooth transition time takes about 22ms in the case
of transiting from 0dB. (Fixed)
Setting of Volume
Default = FFh
Select Address Operational explanation
&h26 [ 7:0 ]
CommandGain
00
01+23.5dB
30
31
32
……
FE
FF
+24dB
0dB
-0.5dB
-1dB
……
-103dB
-∞
44--1166.. MMaaiinn oouuttppuutt bbaallaannccee
Balance can be attenuated, by the step width of 1dB, from the Volume setting value. At the time of switching, smooth
transition is performed. At the time of switching of Balance, smooth transition is performed. The smooth transition time takes
about 22ms. (Fixed)
When measuring the rated output (practical maximum output), it is measured where the total distortion rate (THD+N) is
10%. Clipping with any output amplitude is possible by using of clipper function, for example, the rated output of 10W or 5W
can be obtained by using an amplifier with 15W output.
Please set the &h27[7] at “H” when using of clipper function.
Default = 0
Select Address Value Operational explanation
Clip level is set in the form of higher-order 8 bit&h2A[7:0] and lower-order 8 bit&h2B[7:0].
The volume for sub output can select with 0.5dB step from +24dB to -103dB.
When changing volume, smooth transition is done. Smooth transition duration is required approximately 22ms when it is
from 0dB. (Fixed)
As for sub output balance, it is possible to be attenuated at 1dB step width from volume setting value. When changing,
smooth transition is done.
When changing balance, smooth transition is done. Smooth transition duration is required approximately 22ms. (Fixed)
L/R Balance setting
The case when rated output (practical maximum output) of the television is measured, total harmonic distortion + noise
(THD+N) measures at the place of 10%. It can obtain the rated output of 10W and 5W for example making use of the
amplifier of 15W output, because it is possible to clip with optional output amplitude by using the clipper function.
Please designate &h30 [7] as” H when using sub output clipper
function.
Default = 0
As for clip level, it sets with superior 8 bits &h31 [7: 0] and subordinate 8 bits &h32 [7: 0].
7 bands Parametric Equalizer of main output and of 3 bands Parametric Equalizer of sub output have used the secondary
IIR type digital filter (Bi-quad Filter).
It is possible to set five coefficient 24 bit of b0, b1, b2, a1 and a2 of Bi-quad Filter (-4~+4) directly from an external.
When this function is used, it can do the filter type and frequency setting, Q value (quality factor) setting and gain setting
other than Peaking, Low-Shelf and High-Shelf unrestrictedly.
(Note) five coefficient have the necessity to make below the ±4, there is no read-out function of setting value and an
automatic renewal function of coefficient RAM.
Register for the coefficient transfer of 24bit
Before transferring into coefficient RAM in a lumping, the data is housed in the register for coefficient transfer from the
micro-computer.
Default = 00h
Select Address Operating explanation
&h8D [ 7:0] bit[23:16] which transfers 24 bit coefficient
&h8E [ 7:0] bit[15:8] which transfers 24 bit coefficient
&h8F [ 7:0] bit[7:0] which transfers 24 bit coefficient
It starts to transmit the coefficient of 24bit into coefficient RAM
Default = 0
Select Address Value Operating explanation
&h8C [ 7 ]
0 Coefficient transmission stop
1 Coefficient transmission start
Coefficient number appointment of coefficient RAM
Default = 00h
Select Address Operating explanation
&h8C [ 6:0] Coefficient number appointment of coefficient RAM
Appointment of coefficient number other than 14H↔45H is prohibition
1100.. AAbboouutt aa sseettuupp ooff aa cclloocckk,, aanndd tthhee iinnppuutt ooff aa ccoommmmaanndd
The input of MCLK is decided by combination of three kinds of sampling rates (fs=32kHz, 44.1kHz, 48kHz), and three kinds
of magnifications (128 times, 256 times, 512 times).
Sampling rate(fs)
MCLK clock 32kHz 44.1kHz 48kHz
128fs 4.096MHz 5.6448MHz 6.144MHz
256fs 8.192MHz 11.2896MHz 12.288MHz
512fs 16.384MHz 22.5792MHz 24.576MHz
In order that PLL may multiple the dividing output of MCLK, the dividing ratio of MCLK is not concerned with a sampling rate
like explanation in Chapter 8, but is decided by the magnification of MCLK.
MCLK clock &hF3[5:0]
128fs 04h
256fs 08h
512fs 10h
Therefore, as for the case of the input of 4.096MHz-6.144NHz, and a 256fs setup, in the input frequency of MCLK, in a 128fs
setup, a 16.384MHz - 24.576MHz input serves as a range which can be operated in a 8.192MHz - 12.288MHz input and a
512fs setup.
MCLK
I2C
CONTROL LOGIC
&hF3[5:0]
DIV
PLLAPLL_DIV
DSP
S
E
L
1
AUDIO IF
ERROR_DET
BU9409FV
Clock line
&h08[5:4]
S
E
L
2
The clock system figure of BU9409FV is as mentioned above.
(1) In the case of &h08 [5:4] =1, the block of an above figure light blue operates with a PLL clock.
(2) In the case of &h08 [5:4] =0, the block of an above figure light blue operates by MCLK.
Be careful of the following points at the time of a command input.
In (1), a part of blocks containing DSP are operating with the clock of PLL.
Therefore, even if MCLK is the range which is 4.096MHz - 24.576MHz, when a setup of PLL and the setup of &hF3 are not
performed correctly, a command may not be received other than command &h08 of a system control system, &hA0-&hA9,
&hB0-&hBA, &hD0, &hF0 - &hFA.
In (2), the whole operates with the clock of MCLK.
If MCLK is the range which is 4.096MHz - 24.576MHz, all blocks will receive an I2C command.
hen switching a sampling rate, the clock of the frequency more than the specification range does not go into MCLK, but
when input data is 0, it can return with the following procedures.
1. Carry out the mute of the DAC (MUTEX_SP and MUTEX_DAC are set to L and it is a mute about BD5446.)
↓
○When the input of MCLK has stopped, please do not input a command until MCLK is inputted again.
Please perform the following setup, after MCLK is inputted on the frequency of specification within the limits.
↓
2. It is 20ms or more WAIT because of PLL stability.
↓
○When the section where MCLK stopped or the relation with I2S input had collapsed in the midst of the midst of soft
transition and transmission of a coefficient exists, the coefficient may not be able to be transmitted well. When soft
transition and a coefficient are transmitting, please perform a setup from 11-2 4.
Please perform the following setup, when you are not the midst of soft transition or transmission of a coefficient.
Permanent device damage may occur and break mode (open or short) can not be specified if power supply,
operating temperature, and those of ABSOLUTE MAXIMUM RATINGS are exceeded. If such a special condition is
expected, components for safety such as fuse must be used.
(2)Regarding of SCLI and SDAI terminals
SCLI and the SDAI terminal do not support 5 V-tolerant. Please use it within absolute maximum rating (4.5V).
(3) Power Supply
Power and Ground line must be designed as low impedance in the PCB. Print patterns if digital power supply and
analog power supply must be separated even if these have same voltage level. Print patterns for ground must be
designed as same as power supply. These considerations avoid analog circuits from the digital circuit noise. All pair
of power supply and ground must have their own de-coupling capacitor. Those capacitor should be checked about
their specification, etc. (nominal electrolytic capacitor degrades its capacity at low temperature) and choose the
constant of an electrolytic capacitor.
(4) Functionality in the strong electro-magnetic field
Malfunction may occur if in the strong electro-magnetic field.
(5) Input terminals
All LSI contain parasitic components. Some are junctions which normally reverse bias. When these junctions forward
bias, currents flows on unwanted path, malfunction or device damage may occur. To prevent this, all input terminal
voltage must be between ground and power supply, or in the range of guaranteed value in the Electrical
characteristics. And no voltage should be supplied to all input terminal when power is not supplied.
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