I have use this Alimentation linéaires - Audio-GD PSU-L Alimentation Linéaire discréte + 5V 500mA and this Transformateurs torique - Transformateur torique 15VA 2x12V , my schema
An externally hosted image should be here but it was not working when we last tested it.
usb partagé sur ZimageZ
Where my error?
this way,
http://en.zimagez.com/zimage/untitleda978421837d6514ca196e356fae40448.php
An externally hosted image should be here but it was not working when we last tested it.
http://en.zimagez.com/zimage/untitleda978421837d6514ca196e356fae40448.php
I have a dual mono BIII with Sidecar and Amanero USB module. PCM plays fine through Amanero module, in stereo with R and L channels phased properly. DSD files play in mono. I also converted Left, Right and Both PCM test tracks to DSD and confirmed R and L channel test tracks play from both speakers. The DAC is I2C controlled but no I2C on the USB module. I am using foobar with ASIO Combo384 driver per recent June 2013 HiFiDuino blog Amanero update. No channel swap done in foobar; nothing done in I2C Actions Setup with Amanero config tool.
I have the latest firmware applied to the USB module. Both BIII boards are configured per manual; that is no JPCMx or JDSDx jumpers installed when Sidecar is used. I don't know if this is a configuration issue I am missing with the Amanero USB module or something in the DAC? Appreciate any guidance.
I have the latest firmware applied to the USB module. Both BIII boards are configured per manual; that is no JPCMx or JDSDx jumpers installed when Sidecar is used. I don't know if this is a configuration issue I am missing with the Amanero USB module or something in the DAC? Appreciate any guidance.
I don't think the Combo384 is to blame for that problem, thee is no way it can know to what kind of DAC it is connected to and sends the same sdignal regardless it is a single stereo Buffalo or a dual mono confinguration or any other DAC for that matter.
The problem is most probably lays in the Buffalo or maybe even worse, in the ES9018 chip.
AFAIK it only contains a fix for a function called ASIOfuture that sets up the PCM or DSD format. There was a floating return value. It affected PCM file play in JRiver, it prouced an error when ASIO output mode was used.
thanks for the info, listening to foobar and it gave no improvements in the clicking experience. Started playing a DSD gives a DC, even while playing, pause and continue again it begins with two nasty clicks ... seems there is mouse hiding somewhere in my audio path performing some double clicks and feeling happy ...
thanks for the info, listening to foobar and it gave no improvements in the clicking experience. Started playing a DSD gives a DC, even while playing, pause and continue again it begins with two nasty clicks ... seems there is mouse hiding somewhere in my audio path performing some double clicks and feeling happy ...
Probably the best way to get rid of the click would be using I2C to mute/change format/unmute. If your DAC allows for it try using the Combo384 I2C tool.
I tried the Foobar fix firmware under Foobar with both WASAPI and ASIO and also both in JRiver but the click was still there. As I only listen through HPs and is not that loud it doesn't bother me that much.
That's the same over here, I've got just three real tracks of DSD so I mainly play 192 or 176 so it does not show that much, but it is just anoying while showing off to fellow listeners when they want to hear the tops I can play with my gear .... but the i2c could work ... it's amanero connected to the diyinhk dac .... thanks for the tip ...
The Monster DAC
Last edited:
I agree with you. And I'm not smart enough to figure it out.
I can tell you I got it to work but cannot play SPDIF now. I had to make a new ribbon harness from Sidecar to L and R channel DAC boards. I cut D2 (DSD right channel) at the left channel DAC board and D1 (DSD left channel) at the right DAC board. To get sound from right speaker, I had to connect D2 (DSD right channel) to the D1 input on right channel DAC board (D1 would be left channel input in single BIII DAC application). Now my Right and Left DSD test tracks play from respective speaker. With music vocals are centered, in phase, sounds great.
Wih BIII dual mono, one half of each DAC board is in anti phase making internal DAC mapping complicated for DSD, to me anyway.
Attachments
Last edited:
Hi all,
I have observed a phenomenon with the combo384 that really puzzles me
When playing back wave files (.wav) of everything above 22.050 Hz (fs/2 of the 44.1 kHz basic sampling frequency) with foobar or audacity it seems that the combo384 "mirrors" what is above 22.050 kHz down into what is below 22.050 kHz.
Thus, if I play back a 30 kHz tone via MME or directsound settings in audacity or by selecting amanero in foobar, what is output to my oscilloscope is 22.050 kHz - (30 kHz - 22.050 kHz)= 14.1 kHz ... Both foobar and audacity indicates that the playback frequency/resolution is 192kHz/24 bits so it should come through ...
On the other hand when I output the same 30 kHz tone directly via the wavegene oscillator but instead via wasapi the combo384 sends out 30 kHz to the scope as it should. However, again, when I output the 30 kHz tone from wavegene via MME or direct sound it ends up being 14.1 kHz.
Alternatively, if I output the 30 kHz tone via the computer's own soundcard (realtek something set to a default 192 kHz/24bit resolution) it outputs everything up to 192 kHz/2 as it should.
So I wonder what is happening here & if someone else have observed this? It actually also made me wonder if when playing back 192 kHz/24 bit music files the combo384 actually plays back up to 22050 Hz and then the rest is reflected back into the 0-22050 Hz region ... ? Yet, maybe not easily heard because it's probably low in level ...
A puzzled me,
Jesper
I have observed a phenomenon with the combo384 that really puzzles me
When playing back wave files (.wav) of everything above 22.050 Hz (fs/2 of the 44.1 kHz basic sampling frequency) with foobar or audacity it seems that the combo384 "mirrors" what is above 22.050 kHz down into what is below 22.050 kHz.
Thus, if I play back a 30 kHz tone via MME or directsound settings in audacity or by selecting amanero in foobar, what is output to my oscilloscope is 22.050 kHz - (30 kHz - 22.050 kHz)= 14.1 kHz ... Both foobar and audacity indicates that the playback frequency/resolution is 192kHz/24 bits so it should come through ...
On the other hand when I output the same 30 kHz tone directly via the wavegene oscillator but instead via wasapi the combo384 sends out 30 kHz to the scope as it should. However, again, when I output the 30 kHz tone from wavegene via MME or direct sound it ends up being 14.1 kHz.
Alternatively, if I output the 30 kHz tone via the computer's own soundcard (realtek something set to a default 192 kHz/24bit resolution) it outputs everything up to 192 kHz/2 as it should.
So I wonder what is happening here & if someone else have observed this? It actually also made me wonder if when playing back 192 kHz/24 bit music files the combo384 actually plays back up to 22050 Hz and then the rest is reflected back into the 0-22050 Hz region ... ? Yet, maybe not easily heard because it's probably low in level ...
A puzzled me,
Jesper
I need some help here - I must be doing something wrong :-( I'm trying to flash the Amanero into slave2224 mode, and I can't seem to get it to take (when I pull the clock input pin from MCLK, it continues to play). If I just select slave2224 and CPLD then exit, the device isn't found. If I first flash slave2224 then firmware 074, the device works, but it seems not in slave mode. What am I missing?