I Haven't got it yet. It's just 3300 ohm+1nF cap in parallel like Italians.
When music stars it pops, a lot, when format changes flac ... dff it pops. Gain is much smaller.
Jriver is what I'm using. Foobar2000 sound good with flac but horrible with dff. Hqplayer is next.
Another idea is to use a small form factor low power computer like this:
http://solid-run.com/freescale-imx6-family/hummingboard/
Mount it on the board. Install Audiolinux on the microsd card slot:
http://www.tophifi.it
use the preinstalled Jriver for native DSD 256. Then you use the portable hard drive/NAS to store and stream files, and Jremote with your iPhone for GUI.
There's a better solution. 🙂
Last edited:
It could if you weren't doing any processor heavy things. Or if you used it as an NAA, and did the heavy lifting on a server.
HQplayer , HQplayer embedded, and the NAA daemon is preinstalled on Audiolinux.
Copy image to SD card, power up, and ready to go. Everything you need already installed! It's great.
Not when there's audio hardware to discuss 🙂.
Tapatalk blasts out a loud chime to wake me up 🙂
Just a few thoughts. I've never designed these.
Looks great, do I see correctly that this board will only be connected on one of the JLSound headers, in which case you need some mounting holes.
Keeping the foodprint small is a good thing, but the component spacing is tight. With the variable filter options a bit more space around these components would be sensible I think.
From what I understand keeping the cap in a mm range of the Flipflop would be best. Moving the cap next to the chip might increase the board length but worth it I think.
Also I personally favour having connections on the edge of the board, not in the centre
Why should this (http://www.etalonacoustics.fr/FR/PAGE_Streamer.htm) be better then PIPO X8 TV Box Dual Boot-98.99 and Free Shipping| GearBest.com except that is doesnot have a screen..... Eur 92...
I use something like this, goes very well! (i do not use pcm to dsd conversion!)
Gr GJ
It will run if you're not using it for what it's meant to do. On a low powered computer, best to use as an NAA. But then you need a server running HQplayer. If you want portable, better off using Jriver, and controlling via Jremote. Jriver can run smooth with very little CPU power
That is the burning question. I have no direct interest to run HQplayer at the moment.
As a server in my setup with Foobar and spotify and a usb dac it works and sounds rather good.
Gr GJ
dac= AK449+xmos
You want to use HQplayer for DSD DAC's. This is because it has the best algorithms to resample PCM to DSD of any media player. This way you can get the best quality possible from PCM, using a very simple and good quality DSD DAC.
It's also the best sounding for native DSD music too.
@bbp
Today DAC´s working with silicon (AK, ES, TI and many others) and they have as well to handle PCM.
Native DSD material isn´t really extensively available. I´m sure it will change in near future.
But don´t care, buy HQPlayer and have fun with NoDAC.
My 2 cents, Jean-Paul
Today DAC´s working with silicon (AK, ES, TI and many others) and they have as well to handle PCM.
Native DSD material isn´t really extensively available. I´m sure it will change in near future.
But don´t care, buy HQPlayer and have fun with NoDAC.
My 2 cents, Jean-Paul
@jps1964:
Just out of curiosity: May you have some information about why you are sure this will change? I've been looking into the software and processing available for recording DSD and outside of DSD128 (Korg) and a few other proprietary systems (like e.g. Mergin which uses the AT1201 ADC which is a "6 bit ADC") I have not found much that will allow the path from ADC - in native DSD - into the computer - editing in the computer.
But might there be software available?
@nautibuoy: What really matters is the loop distance between a signal's start until it basically ends the same place it started (hence the "loop distance"). Thus, the 100 nF capacitor has a longer loop distance than the circular capacitor you use just next to the FF. I would reverse the position of these two capacitors. BTW 100 nF has its lowest impedance around 12 MHz - thus it probably would be "optimized" for DSD256 - just FYI in case you didn't consider this ... ? Also, I would again suggest that you take into account the ground bounce phenomenon and to that end also mention that two FFs could improve channel separation. Also, the distance from the output resistors to the output capacitors (including ground return - i.e. the loop distance) should be as short as possible. At these frequencies inductances even in 1 mm length of track may cause HF attenuation - if the HF attenuation happens in the track between the output resistors and the filtering capacitors the HF noise will end up being passed on to the electronics down the line in the audio system.
Best regards,
Jesper
Just out of curiosity: May you have some information about why you are sure this will change? I've been looking into the software and processing available for recording DSD and outside of DSD128 (Korg) and a few other proprietary systems (like e.g. Mergin which uses the AT1201 ADC which is a "6 bit ADC") I have not found much that will allow the path from ADC - in native DSD - into the computer - editing in the computer.
But might there be software available?
@nautibuoy: What really matters is the loop distance between a signal's start until it basically ends the same place it started (hence the "loop distance"). Thus, the 100 nF capacitor has a longer loop distance than the circular capacitor you use just next to the FF. I would reverse the position of these two capacitors. BTW 100 nF has its lowest impedance around 12 MHz - thus it probably would be "optimized" for DSD256 - just FYI in case you didn't consider this ... ? Also, I would again suggest that you take into account the ground bounce phenomenon and to that end also mention that two FFs could improve channel separation. Also, the distance from the output resistors to the output capacitors (including ground return - i.e. the loop distance) should be as short as possible. At these frequencies inductances even in 1 mm length of track may cause HF attenuation - if the HF attenuation happens in the track between the output resistors and the filtering capacitors the HF noise will end up being passed on to the electronics down the line in the audio system.
Best regards,
Jesper
