Hello
I got a Sonic Frontiers SFD2 MK2. Presently, I'm listening music by Qobuz in my Melco N1z 6T connected to an Aufio gd DI-20 interface with the USB yo SPDIF of an Aqua LaVoce Dac.
I simply love the old SFD2 which was refurbished but I can only listen 16/ 44-48 kHz files
I can't select the sampling rate with the interface,
Question:
If I use a chip like the WM8805 to replace the Ultraanalog SPDIF receiver in the SFD2, will I hear high resolution files with this dac truncated to 44 -48 kHz or the Ultraanalog chips which goes to 48 kHz will not let files over 44-48 kHz play ?
Any other solutions ?
Thanks
I got a Sonic Frontiers SFD2 MK2. Presently, I'm listening music by Qobuz in my Melco N1z 6T connected to an Aufio gd DI-20 interface with the USB yo SPDIF of an Aqua LaVoce Dac.
I simply love the old SFD2 which was refurbished but I can only listen 16/ 44-48 kHz files
I can't select the sampling rate with the interface,
Question:
If I use a chip like the WM8805 to replace the Ultraanalog SPDIF receiver in the SFD2, will I hear high resolution files with this dac truncated to 44 -48 kHz or the Ultraanalog chips which goes to 48 kHz will not let files over 44-48 kHz play ?
Any other solutions ?
Thanks
Yes, I know but my goal is not playing files over 24/48, I just want playing 24/96 and 24/192 limited to 24/48.
As is, 24/96 and 24/192 don't play at all with this dac. So if the receiver can do 24/192, will hi Rez files play and be limited to 24/48 by the HDCD chip ?
As is, 24/96 and 24/192 don't play at all with this dac. So if the receiver can do 24/192, will hi Rez files play and be limited to 24/48 by the HDCD chip ?
No. The PMD100 tops out at 55 kHz.
Any choice you make will, potentially, alter the sound of the dac. Your could precede the PMD100 with a sample rate converter which will convert any input sample rate to a fixed output sample rate chosen by you. The other choice is to replace the PMD100 with a 96k/192k capable digital filter.
Any choice you make will, potentially, alter the sound of the dac. Your could precede the PMD100 with a sample rate converter which will convert any input sample rate to a fixed output sample rate chosen by you. The other choice is to replace the PMD100 with a 96k/192k capable digital filter.
Sample rate converter seems the easy way to do this task. I really don't know anything about this. Is there any serious SC so I will not affect the sound quality ? In fact, most of the files I have (4T) are 16/44 and half of the Qobuz files I have are also 16/44. Listening 16/44 files on the SFD2 Mk2 gives me more pleasure than listening them on my Aqua LaVoce S3.
Foobar2000 has several options for high quality resampling. Should be quite easy to setup with SFD2 Mk2 if your computer has a SPDIF or optical output.
I'm using a Melco streamer N1Z 6T not a computer. Computer in the past gave me more possibilities than a streamer but the sound of a streamer is in another league.
Hi think you need df1706 filter upgrade:
https://www.ebay.it/itm/37344062133...b-EMIomQxa&var=&widget_ver=artemis&media=COPY
Plus new receiver , please see pictures
https://www.ebay.it/itm/37344062133...b-EMIomQxa&var=&widget_ver=artemis&media=COPY
Plus new receiver , please see pictures
Attachments
Hello Tonixp
Great. Now where can I find the SPDIF module to replace the AES21 module ?
I could perhaps use I2s to bypass SPDIF module.
Great. Now where can I find the SPDIF module to replace the AES21 module ?
I could perhaps use I2s to bypass SPDIF module.
Given that the the majority of your listening is 16/44, in your position, I would not want to change the SFD2 in any way. And the DF1706 is a very different animal to the PMD100. I would make my changes outside the SFD2.
Not sure about the Taobao option though.
https://www.diyaudio.com/community/threads/ultra-analog-aes-21-upgrade-replacement-project.250700/
Not sure about the Taobao option though.
As I really love the sound as it is, making changes outside could be the best.
I think I could listen Qobuz with UPNP on the Melco and settle the frequency to 44 kHz.
I think I could listen Qobuz with UPNP on the Melco and settle the frequency to 44 kHz.
1) The socket uses very high quality parts, the best that I have found on the market for every SMD/TH parts.
For the wm8804 ic , I order genuine parts from a reputable vendor (they are of course expensive).
The resultate is very nice!
But the problem now is that I don't have ago the equipment to test the new sockets because the Parasound DAC 2000 it come back to Holland .
So as soon as i can , i will come back on this item for project also a check control quality pcb .
Please stay tuned .
2) Yes you can also bypass the spdf module but you need a 4 wire PCM format signal.
LRCK
BCK
DATA LEFT
DATA RIGHT
Also a RJ45 or HDMI connector fixed on rear panel and a appropriate source by the way.
My interface has a HDMI output in balanced mode so I assume I use only the + signals. Here is the output of the interface. Which is DATA R and DATA L
This Is a balanced i2s format.
It Is used to transport the i2s from source to D20400 DAC module.
Next step Is to convert this format in PCM format.
It Is possibile but complicate becouse your need also this:
https://www.audiophonics.fr/en/interface-modules/ian-canada-i2s-to-pcm-conversion-board-p-16964.html
So Is better the first solution.
It Is used to transport the i2s from source to D20400 DAC module.
Next step Is to convert this format in PCM format.
It Is possibile but complicate becouse your need also this:
https://www.audiophonics.fr/en/interface-modules/ian-canada-i2s-to-pcm-conversion-board-p-16964.html
So Is better the first solution.
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In my interface, there's unbalanced signals that goes with 47 ohms to a chip which balanced to the output.
I tried the unbalanced output without success with something else. Maybe I could use this again to feed the PMD100 ?
I tried the unbalanced output without success with something else. Maybe I could use this again to feed the PMD100 ?
The most problem of i2s unbalanced trasporto Is the lengh of cable (max 30 cm).
So :
1)you have to continue to use balanced type.
2) you have to convert in pcm format and bypassing digital filter you will obtain the max bit and sampling rate performances of AD20400 that i don't know.
So :
1)you have to continue to use balanced type.
2) you have to convert in pcm format and bypassing digital filter you will obtain the max bit and sampling rate performances of AD20400 that i don't know.
If I take the 2e solution, 24/96 and 24/192 files will play at 20/48 as the AD20400 can do or there's a risk I will have no sound ?
This will depend on which DAC chip is present inside (i don't know) the AD20400 and consequently on its technical specifications.
For example, the TDA1541, although implemented in 99% of cases in I2S mode and coupled with its SAA7220 filter, cannot work beyond 16bit/44.1khz, but things change if you set it in SIM and NOS mode, in this case it can work easily at 24bit/192Khz.
For more info about AD20400 you can see relative THD.
For example, the TDA1541, although implemented in 99% of cases in I2S mode and coupled with its SAA7220 filter, cannot work beyond 16bit/44.1khz, but things change if you set it in SIM and NOS mode, in this case it can work easily at 24bit/192Khz.
For more info about AD20400 you can see relative THD.
Today, I listened again 16/44 files on the SFD2 MK2 and on the Aqua LaVoce S3. For my taste, the SFD2 MK2 wins the contest. I love it so much I could forget listening hi-Rez files and listening them at 16/44 on the SFD2.
The SFD2 has more 'Meat' timbre. Sound simply has more real life. If I could find something I only have to transform files instead of modifying the dac it could be great.
The SFD2 has more 'Meat' timbre. Sound simply has more real life. If I could find something I only have to transform files instead of modifying the dac it could be great.