Re: Never seen this??
Errr.... what is it?
koldby said:
Wonder if anybody here has seen this:
http://listen.to/audioexperiment
I know this is not a digital in amp, but it sure is a DIY project!!
Errr.... what is it?
There may not be any point in reviving this thread, but I recently became very interested in the Panny XR45, initially due to this editorial from the Newform site:
http://www.newformresearch.com/updateaug03.htm
After doing a bit of research (including this thread), I'm very interested some ideas similar to what Brian is doing. Specifically, I'm interested in trying to tap into the I2S signals to/from the equibit processors and bridging them into an I2S based soundcard like the Envy24 cards. This would allow both direct digital output of the multi-channel signal to the PC, as well as direct digital output from the PC into the amp sections.
I (of course) know just enough about all this to be dangerous, but any updates on experiences with the Panasonic units or other equibit (or other) amps along these lines would be greatly appreciated.
http://www.newformresearch.com/updateaug03.htm
After doing a bit of research (including this thread), I'm very interested some ideas similar to what Brian is doing. Specifically, I'm interested in trying to tap into the I2S signals to/from the equibit processors and bridging them into an I2S based soundcard like the Envy24 cards. This would allow both direct digital output of the multi-channel signal to the PC, as well as direct digital output from the PC into the amp sections.
I (of course) know just enough about all this to be dangerous, but any updates on experiences with the Panasonic units or other equibit (or other) amps along these lines would be greatly appreciated.
SA-XR10 Update
I’ve done some more investigation and experimenting with my SA-XR10. Here’s some new information that I can share:
The main Equibit amplifier board appears to be made by Texas Instruments. It seems that Panasonic just built the rest of the receiver around it. This board is pretty well self-contained. I’m able to operate it independently outside of the receiver. The layout is very similar to TI’s other Equibit reference boards (and it’s much cheaper to buy the receiver).
Panasonic sells the service manual for this receiver. It includes full schematics.
Order number: AD0207128C1
1-800-833-9626
They only charge about $7 for it. It’s fully worth the price for the detail that it provides, rather than try to make copies.
I can offer the following supplemental information:
Most of the connections to it are made with ZIF connectors. The bare end of stripped wire can be inserted into them and locked into place.
These include:
Five Speaker Outputs (channels 1-5) (the receiver adds some snubber caps at the final output terminals).
Subwoofer line out (an op-amp circuit integrates the channel 6 Equibit PWM back into a line signal).
Interface to headphone output circuit board (can be left unconnected if using the board outside the receiver).
Main bus power (45VDC nominal).
The main input to this board is through a 25pos 1mm FPC connector.
It includes the digital audio interface, status/control lines, amplifier power supply control, and low level power.
PINOUT <direction> NAME (description):
1. <digital_in> MCLK (256fs)
2. <digital_in> SCLK (64fs)
3. <digital_in> LRCLK (fs)
4. DGND
5. <digital_in> FRONT DATA (I2S 24bit data)
6. <digital_in> SURROUND DATA (I2S 24bit data)
7. <digital_in> CENTER / SUB DATA (I2S 24bit data)
8. DGND
9. <analog_out> DEC (Equibit bus voltage feedback) (DEC = BUS / 11)
10. DACGND (DGND)
11. <digital_in> DOUBLE SPEED (for audio data rate) (0: fs = 44.1KHz or 48KHz, 1: fs = 88.2KHz or 96KHz)
12. <digital_out> /TEMP WARNING (0: High heat sink temp – amp still functions, 1: Normal)
13. <analog_in> VOLUME (Equibit bus voltage control) (BUS = VOLUME * 13.3)
14. <digital_out> /Shutdown (0: Overtemp or output short – amp disabled, 1: Normal)
15. <analog_in> AMP RESET (see note)
16. AC GND (DGND)
17. <digital_in> /MUTE – HeadPhones (0: Headphones disabled, 1: Enabled)
18. <digital_in> /MUTE – Front (0: Front speakers disabled, 1: Enabled)
19. <digital_in> /MUTE – Center & Surround (0: Center and surround speakers disabled, 1: Enabled)
20. <digital_in> /MUTE – Subwoofer (0: Subwoofer line out disabled, 1: Enabled)
21. 12V_GND (DGND)
22. +B (12V)
23. 12V_GND (DGND)
24. +B (12V)
25. <digital_out> Headphone Switch (0: Headphones plugged in, 1: No headphones)
NOTES:
Logic signals are 3.3V (LVTTL).
This board derives its internal 3.3V supply from the external +B (12V) supply.
The AMP RESET input is not a logic control signal. It is a +5V supply connection to a power monitor chip. It is intended to be connected to an external +5V supply that is derived from the same +B (12V) supply that feeds this board.
************
This board is based on three stereo TAS5012 Equibit modulator chips. The power output stage is a discreet implementation. It has five channels of amplification. The sixth (subwoofer) channel goes through the same Equibit modulator section as the other five channels, but instead of proceeding through a switching power output stage, it gets integrated into a line level output.
The board also has a switching power supply to provide the variable-voltage DC bus for the Equibit output power H-bridges. This is based on a UC3849DW Secondary Side Average Current Mode Controller. This power supply sets the bus voltage at 13.3 times the voltage input to the board’s VOLUME input. The bus voltage can be verified by reading the voltage of the DEC output (bus voltage divided by 11). Lowering the bus voltage decreases the output volume without reducing the 24bit audio resolution.
Variable bus voltage can achieve 20dB of output level variation. The variation is limited to 20dB because of two constraints: The minimum voltage is limited by the voltage drop of the output MOSFETs and the output low-pass filter. The maximum voltage is limited by the voltage rating of the output MOSFETs and/or the gate driver circuit. The SA-XR10 has a bus voltage range of 4.33V to 42.2V (20dB).
The output volume is controlled by a combination of digital attenuation and variable bus supply voltage. This board requires external control of the bus voltage and some type of digital attenuation before data is input into it.
The scheme the SA-XR10 receiver uses is as follows:
0dB (full output): no digital attenuation and full bus voltage.
Note that (assuming the data isn’t clipped digitally) this amplifier can’t be driven into clipping. (It is possible to cause the thermal shutdown circuit to activate, however.)
The first 20dB of volume control (-1dB to -20dB) are done digitally. (At -20dB, the digital resolution is effectively reduced from 24bits to a little over 20bits).
The second 20dB of volume control (-21dB to -40dB) are done by reducing the bus voltage. (The effective resolution remains at about 20bits).
The remaining 39dB of volume control (-41dB to -79dB) are again done digitally. (At -79dB, the effective resolution decreases to about 14 bits – pretty good considering that the peak levels at this setting are really quiet.)
Below -79dB the output is muted.
To operate this board without the rest of the receiver, you have to supply your own combination of digital attenuation and/or level control.
A very crude initial method that I’ve been successful with is to shift the input data right 3 bits (-18dB) with a CPLD and then use a pot to control the VOLUME supply control input. I use a 10K log taper pot with the top side connected to 3.3V through a 453ohm resistor and the low side connected to ground through a 1.13K resistor. This produces a log taper control voltage at the wiper with a range of 0.3V to 3.2V, which corresponds to a supply voltage range of about 4V to 42V. This simple control method lets me have a range of -18dB to -38dB, which has been completely adequate.
*********
I view the variable switching power supply as the weak point of this board’s design. Watching the bus with a scope, things are pretty clean at lower levels. At high output levels things start to get pretty wiggly. The Equibit output section basically has no power supply rejection, so this is an issue. (I’m sure this is the reason for the SA-XR10’s published 0.9% THD at 100W, it will be drastically lower at reduced power levels.) The Equibit design really needs a stiff, low-impedance supply even more than most other amps. As an initial experiment I isolated the bus from the switching supply and hooked it up to three 12V car batteries in series. This made an astounding improvement at loud levels. I’ve already mentioned the surprising clarity and imaging this unit had at more moderate levels. With the batteries, percussion was especially improved.
**********
I’m still concentrating on a multichannel digital interface for DVD-A and SACD at the moment. I’ve been doing some experiments with different clocking and reclocking schemes. I’m not ready to report on this yet, but I’ve made some good progress. I’ll (hopefully) be getting back to my Equibit experiments soon.
(Did anybody else notice that Digikey now has TAS5182 chips available?! TI hasn’t even been sampling them yet.)
*********
A couple of comments about the SA-XR25 and SA-XR45:
I haven’t seen or heard either one of these in person, much less probed around inside of one.
I have reviewed some of their documentation.
It appears that both have identical Equibit sections.
There are some major differences from the SA-XR10:
The Equibit board also has a local processor that supports other receiver functions.
The bus supply is on a separate board and operates at a single voltage.
Digital attenuation is done in combination with a TAS5036 six channel Equibit modulator chip (with built in volume control).
There are six power output stages instead of five.
The output stages are now based on the TAS5182 chip, instead of discreet components.
The output MOSFETs and low pass filters appear to be identical to the SA-XR10.
I’m not surmising that these are any better or worse in audio quality than the SA-XR10. I just wanted to point out that interfacing to the internal boards will be quite different from what I’ve reported above.
**********
I’ll post some more stuff later as I progress with my (slow) progress.
Regards,
Brian.
I’ve done some more investigation and experimenting with my SA-XR10. Here’s some new information that I can share:
The main Equibit amplifier board appears to be made by Texas Instruments. It seems that Panasonic just built the rest of the receiver around it. This board is pretty well self-contained. I’m able to operate it independently outside of the receiver. The layout is very similar to TI’s other Equibit reference boards (and it’s much cheaper to buy the receiver).
Panasonic sells the service manual for this receiver. It includes full schematics.
Order number: AD0207128C1
1-800-833-9626
They only charge about $7 for it. It’s fully worth the price for the detail that it provides, rather than try to make copies.
I can offer the following supplemental information:
Most of the connections to it are made with ZIF connectors. The bare end of stripped wire can be inserted into them and locked into place.
These include:
Five Speaker Outputs (channels 1-5) (the receiver adds some snubber caps at the final output terminals).
Subwoofer line out (an op-amp circuit integrates the channel 6 Equibit PWM back into a line signal).
Interface to headphone output circuit board (can be left unconnected if using the board outside the receiver).
Main bus power (45VDC nominal).
The main input to this board is through a 25pos 1mm FPC connector.
It includes the digital audio interface, status/control lines, amplifier power supply control, and low level power.
PINOUT <direction> NAME (description):
1. <digital_in> MCLK (256fs)
2. <digital_in> SCLK (64fs)
3. <digital_in> LRCLK (fs)
4. DGND
5. <digital_in> FRONT DATA (I2S 24bit data)
6. <digital_in> SURROUND DATA (I2S 24bit data)
7. <digital_in> CENTER / SUB DATA (I2S 24bit data)
8. DGND
9. <analog_out> DEC (Equibit bus voltage feedback) (DEC = BUS / 11)
10. DACGND (DGND)
11. <digital_in> DOUBLE SPEED (for audio data rate) (0: fs = 44.1KHz or 48KHz, 1: fs = 88.2KHz or 96KHz)
12. <digital_out> /TEMP WARNING (0: High heat sink temp – amp still functions, 1: Normal)
13. <analog_in> VOLUME (Equibit bus voltage control) (BUS = VOLUME * 13.3)
14. <digital_out> /Shutdown (0: Overtemp or output short – amp disabled, 1: Normal)
15. <analog_in> AMP RESET (see note)
16. AC GND (DGND)
17. <digital_in> /MUTE – HeadPhones (0: Headphones disabled, 1: Enabled)
18. <digital_in> /MUTE – Front (0: Front speakers disabled, 1: Enabled)
19. <digital_in> /MUTE – Center & Surround (0: Center and surround speakers disabled, 1: Enabled)
20. <digital_in> /MUTE – Subwoofer (0: Subwoofer line out disabled, 1: Enabled)
21. 12V_GND (DGND)
22. +B (12V)
23. 12V_GND (DGND)
24. +B (12V)
25. <digital_out> Headphone Switch (0: Headphones plugged in, 1: No headphones)
NOTES:
Logic signals are 3.3V (LVTTL).
This board derives its internal 3.3V supply from the external +B (12V) supply.
The AMP RESET input is not a logic control signal. It is a +5V supply connection to a power monitor chip. It is intended to be connected to an external +5V supply that is derived from the same +B (12V) supply that feeds this board.
************
This board is based on three stereo TAS5012 Equibit modulator chips. The power output stage is a discreet implementation. It has five channels of amplification. The sixth (subwoofer) channel goes through the same Equibit modulator section as the other five channels, but instead of proceeding through a switching power output stage, it gets integrated into a line level output.
The board also has a switching power supply to provide the variable-voltage DC bus for the Equibit output power H-bridges. This is based on a UC3849DW Secondary Side Average Current Mode Controller. This power supply sets the bus voltage at 13.3 times the voltage input to the board’s VOLUME input. The bus voltage can be verified by reading the voltage of the DEC output (bus voltage divided by 11). Lowering the bus voltage decreases the output volume without reducing the 24bit audio resolution.
Variable bus voltage can achieve 20dB of output level variation. The variation is limited to 20dB because of two constraints: The minimum voltage is limited by the voltage drop of the output MOSFETs and the output low-pass filter. The maximum voltage is limited by the voltage rating of the output MOSFETs and/or the gate driver circuit. The SA-XR10 has a bus voltage range of 4.33V to 42.2V (20dB).
The output volume is controlled by a combination of digital attenuation and variable bus supply voltage. This board requires external control of the bus voltage and some type of digital attenuation before data is input into it.
The scheme the SA-XR10 receiver uses is as follows:
0dB (full output): no digital attenuation and full bus voltage.
Note that (assuming the data isn’t clipped digitally) this amplifier can’t be driven into clipping. (It is possible to cause the thermal shutdown circuit to activate, however.)
The first 20dB of volume control (-1dB to -20dB) are done digitally. (At -20dB, the digital resolution is effectively reduced from 24bits to a little over 20bits).
The second 20dB of volume control (-21dB to -40dB) are done by reducing the bus voltage. (The effective resolution remains at about 20bits).
The remaining 39dB of volume control (-41dB to -79dB) are again done digitally. (At -79dB, the effective resolution decreases to about 14 bits – pretty good considering that the peak levels at this setting are really quiet.)
Below -79dB the output is muted.
To operate this board without the rest of the receiver, you have to supply your own combination of digital attenuation and/or level control.
A very crude initial method that I’ve been successful with is to shift the input data right 3 bits (-18dB) with a CPLD and then use a pot to control the VOLUME supply control input. I use a 10K log taper pot with the top side connected to 3.3V through a 453ohm resistor and the low side connected to ground through a 1.13K resistor. This produces a log taper control voltage at the wiper with a range of 0.3V to 3.2V, which corresponds to a supply voltage range of about 4V to 42V. This simple control method lets me have a range of -18dB to -38dB, which has been completely adequate.
*********
I view the variable switching power supply as the weak point of this board’s design. Watching the bus with a scope, things are pretty clean at lower levels. At high output levels things start to get pretty wiggly. The Equibit output section basically has no power supply rejection, so this is an issue. (I’m sure this is the reason for the SA-XR10’s published 0.9% THD at 100W, it will be drastically lower at reduced power levels.) The Equibit design really needs a stiff, low-impedance supply even more than most other amps. As an initial experiment I isolated the bus from the switching supply and hooked it up to three 12V car batteries in series. This made an astounding improvement at loud levels. I’ve already mentioned the surprising clarity and imaging this unit had at more moderate levels. With the batteries, percussion was especially improved.
**********
I’m still concentrating on a multichannel digital interface for DVD-A and SACD at the moment. I’ve been doing some experiments with different clocking and reclocking schemes. I’m not ready to report on this yet, but I’ve made some good progress. I’ll (hopefully) be getting back to my Equibit experiments soon.
(Did anybody else notice that Digikey now has TAS5182 chips available?! TI hasn’t even been sampling them yet.)
*********
A couple of comments about the SA-XR25 and SA-XR45:
I haven’t seen or heard either one of these in person, much less probed around inside of one.
I have reviewed some of their documentation.
It appears that both have identical Equibit sections.
There are some major differences from the SA-XR10:
The Equibit board also has a local processor that supports other receiver functions.
The bus supply is on a separate board and operates at a single voltage.
Digital attenuation is done in combination with a TAS5036 six channel Equibit modulator chip (with built in volume control).
There are six power output stages instead of five.
The output stages are now based on the TAS5182 chip, instead of discreet components.
The output MOSFETs and low pass filters appear to be identical to the SA-XR10.
I’m not surmising that these are any better or worse in audio quality than the SA-XR10. I just wanted to point out that interfacing to the internal boards will be quite different from what I’ve reported above.
**********
I’ll post some more stuff later as I progress with my (slow) progress.
Regards,
Brian.
I'm using the SA-XR10 now in a fully digital i2s connected chain (transport -> ultracurve -> sa-xr10) with full range horn loaded Stax electrostatics (50-20k) on the LR channels and an 18" ELF woofer on the sub channel (10Hz-50Hz). In this setup the built in cyrus decoder vorks fine as a 2 way crossover.
I'm also thinking about power supply modifications or battery supply but I have a question.
Could you figure out how the volume control works on the sub and headphone channel. They are both integrated back from the output of the equibit chips and according to your report the digital volume control made on the DSP board does not shift the bits in the -20 - -40dB range. But it seems both channel follows the volume control even in this range.
Do you have an idea what is the max supply voltage the unit can handle?
Thanks
I'm also thinking about power supply modifications or battery supply but I have a question.
Could you figure out how the volume control works on the sub and headphone channel. They are both integrated back from the output of the equibit chips and according to your report the digital volume control made on the DSP board does not shift the bits in the -20 - -40dB range. But it seems both channel follows the volume control even in this range.
Do you have an idea what is the max supply voltage the unit can handle?
Thanks
Seconded - thanks for the info.
I opened my XR25 up, and it does look rather different from what you describe. The input/decoding board looks pretty densely packed with stuff, but it *looks* like the interface down to the Equibit board might be pretty straightforward.
One big-ish difference is probably going to be in the sub channel handling. Since there are 6 powered output channels in the XR25/45, the sub does not appear to go through an Equibit stage, although from a casual observation I can't see how it is handled.
I have two goals - short term and long term. Long term, I want to get to a full 6-channel I2S interface. In the short term, though, I'd be happy just getting use of all 6 channels from an external analog in.
Brian, any suggestions on where to get the order number for the XR25/45 service manuals. I suppose I can just call the number and wing it.
I opened my XR25 up, and it does look rather different from what you describe. The input/decoding board looks pretty densely packed with stuff, but it *looks* like the interface down to the Equibit board might be pretty straightforward.
One big-ish difference is probably going to be in the sub channel handling. Since there are 6 powered output channels in the XR25/45, the sub does not appear to go through an Equibit stage, although from a casual observation I can't see how it is handled.
I have two goals - short term and long term. Long term, I want to get to a full 6-channel I2S interface. In the short term, though, I'd be happy just getting use of all 6 channels from an external analog in.
Brian, any suggestions on where to get the order number for the XR25/45 service manuals. I suppose I can just call the number and wing it.
For the subwoofer, the volume control is done strictly by digital attenuation, even in the -21dB to -40dB range.fcserei said:
The headphones are integrated from the same Equibit PWM stream as the front speakers. The SA-XR10 can't drive Speakers and Headphones at the same time.
Please note that the Headphone Switch (Pin 25) description I gave in my previous post was backwards. It should have read 0: No headphones, 1: Headphones plugged in.
When *no* headphones are plugged in, the Headphone Switch signal is low. The receiver then pulls the /MUTE - Headphones line low (disabling the headphone integrators). The /MUTE – Front line is high (enabling the power MOSFET stage for the front speakers). Volume control is done as I previously described.
When headphones *are* plugged in, the Headphone Switch signal is high. The receiver then pulls the /MUTE - Front line low (disabling the power MOSFET stage for the front speakers). The /MUTE - Headphones line is high (enabling the headphone integrators). In this state volume control is done by digital attenuation only.
The IRFIZ24N output MOSFETs are rated at 100V. The flyback voltage that these devices see could peak out at double the bus voltage. Also, the bus filter caps are rated at 50V. These two factors would indicate a maximum of 50V. Personally, I wouldn't try to go higher than the 42.2V volts originally designed into the unit.
The real limiting factor of this output stage appears to be the thermal dissipation of the MOSFETs. I don't think a higher bus voltage would be of value unless a completely new output stage was designed.
Brian.
On this unit the sub line outputs and the headphone outputs are handled by conventional A/Ds. They are completely separate from the Equibit section.dwk123 said:
The manual for the SA-XR25 is order number MD0302055C1.
I don't know the number for the SA-XR45, but you should be able to get it from Panasonic Customer Service (1-800-833-9626) by asking for it by description.
Brian.
The XR10 sounds like a great platform for tweaking, the newform research link above shows the potential of this unit.
Interesting that several web based reviews give the unit an average score for sound quality, and that the unit seems to run out of puff at higher levels compared to conventional amps. Perhaps this is due to the problems Brian has outlined in the SMPS at high volumes, which might be fixable.
I was considering ordering a few of the TI eval boards but buying a used XR10 on ebay or audiogon & hacking it looks like a more effective option.
Brian, can you tell if the power supply is switchable from 110v to 220 or 240v? Keep those posts coming, there are several of us on the forum keen to track your progress.
Regards,
Dean
Interesting that several web based reviews give the unit an average score for sound quality, and that the unit seems to run out of puff at higher levels compared to conventional amps. Perhaps this is due to the problems Brian has outlined in the SMPS at high volumes, which might be fixable.
I was considering ordering a few of the TI eval boards but buying a used XR10 on ebay or audiogon & hacking it looks like a more effective option.
Brian, can you tell if the power supply is switchable from 110v to 220 or 240v? Keep those posts coming, there are several of us on the forum keen to track your progress.
Regards,
Dean
deandob said:
My SA-XR10 is a U.S. model. Its main power supply will only run off of 120VAC. It doesn't have a universal input range.
I believe that there were other versions of this unit in other countries for 240VAC. My documentation only covers the U.S. version.
I think they based both versions on the same board, but it would take some time to work out the details.
The main supply provides ~45VDC for the Equibit board, which has its own variable voltage switcher for the bus. If you only want to use the Equibit board by itself, then it probably doesn't matter which version you get.
Brian.
driving 1 ohm loads with digital amp
I want to buiild a digital amp that can safely drive 0.7 to 1.0 ohm speaker loads with 25-50 watts, and deliver high quality sound with very low noise. Any suggestions on modifications to the TI or Tripath, or ?? designs for this low impedance?
Is there more to the design than solving:
Robust power supply of about 24 volts.
Very low Ron Mosfets. Trade-off lower VDD_max for lower Ron.
Driving many parallel output Mosfets.
I want to buiild a digital amp that can safely drive 0.7 to 1.0 ohm speaker loads with 25-50 watts, and deliver high quality sound with very low noise. Any suggestions on modifications to the TI or Tripath, or ?? designs for this low impedance?
Is there more to the design than solving:
Robust power supply of about 24 volts.
Very low Ron Mosfets. Trade-off lower VDD_max for lower Ron.
Driving many parallel output Mosfets.
Brian,
Thanks for the clarification.
What I really don't get, why are they decided to drop bits first in the volume control and then decrease the supply voltage. The S/N ratio is not that great that you can drop 4 bits easily without any sonic penalties. It would be more logical to reduce the voltage first for the 0- -20dB range and then drop resolution bits, especially as you wrote, the variable supply is not noise free at higher voltages.
Most of my listening is happening in the -10 - -25 dB range so I'm thinking to short R621 , thus reducing the multiplication coefficient of the variable supply. This way I'd use the 0- -15dB range with about 15V supply voltage without resolution loss at high volumes and better supply regulation.
What do you think?
Thanks for the clarification.
What I really don't get, why are they decided to drop bits first in the volume control and then decrease the supply voltage. The S/N ratio is not that great that you can drop 4 bits easily without any sonic penalties. It would be more logical to reduce the voltage first for the 0- -20dB range and then drop resolution bits, especially as you wrote, the variable supply is not noise free at higher voltages.
Most of my listening is happening in the -10 - -25 dB range so I'm thinking to short R621 , thus reducing the multiplication coefficient of the variable supply. This way I'd use the 0- -15dB range with about 15V supply voltage without resolution loss at high volumes and better supply regulation.
What do you think?
fcserei said:
I have been thinking of the volume control question as well, particularly in light of the fact that on my XR25 (which uses ONLY digital attenuation, no variable supply) I _still_ get incredible low-level resolution even when listening at -35 or even lower. This is not what I would have expected from a unit with only ~96dB S/N running 35dB below full-scale.
I think the answer has to lie in understanding the noise/distortion mechanisms in the Equibit chipset, and unfortunately I don't really know enough about how they work internally to know. That won't stop me from speculating, though
If the noise/distortion is direcly related to uncertainty/jitter in the PWM master clock, then it will be largely independent of signal level, and will behave like a normal analog amp in that the noise floor will be pretty constant.
However, if the noise/distortion is somehow proportional to the input level (ie percentage inaccuracies in the PWM signal or switching times), then the noise floor would drop with lower signals, and so the unit would preserve that 96 db S/N performance even when operating below full-scale.
Remember that the unit isn't really 'dropping bits', it's only 'shifting bits' since the input to the amps will (presumably) preserve 24 bit word length. This operation by itself won't hurt anything - it all comes down the the noise characteristics of the amp section.
Unfortunately, the most logical source of noise would be jitter in the pwm clock or other inaccuracies in the on/off switching of each pulse, which would seem to be more 'absolute' than 'proportional' in nature, since they would be related to the on/off transitions and not the width of the pulse.....
dwk123 said:
I think the answer has to lie in understanding the noise/distortion mechanisms in the Equibit chipset, and unfortunately I don't really know enough about how they work internally to know. That won't stop me from speculating, though
Where could we learn more about the Equibit coding method. I've never seen any details about the actual modulation.
dwk123 said:
Yes, this is the question. If it is 'absolute', digital volume control is OK, if 'proportional, then shifting bits is bad because of the limited resoulution of the amp.
Anyway. I'll try to reduce the supply voltage gain and check the effect.
It's not that the supply is noiser at higher voltages. Rather it's that the supply is noiser at higher current loads.fcserei said:
If the supply was very stiff and low impedance in the audio frequency range, then it would probably be best to do the initial -1dB to -20dB attenuation by decreasing the bus voltage.
I suspect that with the SA-XR10's particular supply, TI found that it was less noisy to have a higher voltage with a lower current load for the -1dB to -20dB region.
I plan to do some more experimentation in this area.
For the reason I stated above, I don't think it would be beneficial to drop the bus voltage.
Also, there might be a problem because the SA-XR10's control processor probably expects to see a verification of the bus voltage at the DEC pin. A mismatch might cause the receiver to shut down with a fault condition. You could give it a try and see what happens.
This problem could be overcome by rescaling the DEC feedback resistors (change R634 from 27K to 8.45K). This rescaling could possibly cause the DEC signal to overvoltage the A/D converter, so a diode from DEC to +5V (RESET pin) to clamp the signal would be a good idea if you wanted to try this experiment.
Brian.
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