I use 2xMeanwell switch mode -> Hypex HNR12/HPR12 -> Keantokens Multiplier
RS-15-24 Mean Well | Mouser
https://www.hypexshop.com/DetailServlet?detailID=2014
The K Multiplier
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I have the same problem with my RaPi connected to Soekris via I2S.
When I use a Hifiberry Digi connected over SPDIF to Soekris everything was o.k.
🙁
Daniel
I think it will be god idea if everybody who ad 16 capacitors mod on the back of board write what change in sound, and what capacitors was used. Some body says that is something wrong in bas. IT will be easier to decide to do this mod. Very thanks !
Nothing wrong with bass here 🙂
I think the major culprit for bad bass is the series resistor of the original vref PS,
on the V1 boards it's 100 ohm, Soekris recommends it to change to 0.1 ohm and the compensation cap to a 499ohm resistor.
My mods so far on the on the vref PS: extra 47uf on top of existing 22uf SMT,
0.01 ohm series resistor, 499 ohm compensation cap resistor, 9x 680uf polymer (UCC PSA + Nichicon PLE), 2x Wima MKS2 6.8uf, 1x 10nF polystyrene.
That's the 52 cap mod 😉
Mods on the clock: Building towers on the 2 existing 4.7uf caps around ring of clock: added 10nf C0G, 100pf C0G, underneath board through vias: 47uf X5R + 390uf UCC PSA.
So close to the clock there are 2 towers of 5x caps (see pics).
On J9 PS clock added 680 uf Nichicon PLE
Mods +5V: replaced original caps with UCC PSA 270uF
Result of all this: the DAC sounds very very good, plenty of 3D 🙂
Better than my es9018 and definitely on same level as AYA II.
Regards,
Danny
I think the major culprit for bad bass is the series resistor of the original vref PS,
on the V1 boards it's 100 ohm, Soekris recommends it to change to 0.1 ohm and the compensation cap to a 499ohm resistor.
My mods so far on the on the vref PS: extra 47uf on top of existing 22uf SMT,
0.01 ohm series resistor, 499 ohm compensation cap resistor, 9x 680uf polymer (UCC PSA + Nichicon PLE), 2x Wima MKS2 6.8uf, 1x 10nF polystyrene.
That's the 52 cap mod 😉
Mods on the clock: Building towers on the 2 existing 4.7uf caps around ring of clock: added 10nf C0G, 100pf C0G, underneath board through vias: 47uf X5R + 390uf UCC PSA.
So close to the clock there are 2 towers of 5x caps (see pics).
On J9 PS clock added 680 uf Nichicon PLE
Mods +5V: replaced original caps with UCC PSA 270uF
Result of all this: the DAC sounds very very good, plenty of 3D 🙂
Better than my es9018 and definitely on same level as AYA II.
Regards,
Danny
Attachments
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Thanks Daniel for your information.
Your solution is a bit workaround, but it would be nice to fasten RPi to dam1021 I2S without additional equipment. Another workaround is use Sox with mpd in the RPi - I've already checked - upsampling works with 48kHz,88.2kHz,96kHz (but not with 176.4 and higher - RPi is not strong enough). Maybe someone already had this problem and solved it? I've seen posts in this tread from guys they use RPi & I2S with dam1021, so they should discover this, or just this 0.99 firmware has bug.
Maybe Soren can help?
Regards
Jacek
Your solution is a bit workaround, but it would be nice to fasten RPi to dam1021 I2S without additional equipment. Another workaround is use Sox with mpd in the RPi - I've already checked - upsampling works with 48kHz,88.2kHz,96kHz (but not with 176.4 and higher - RPi is not strong enough). Maybe someone already had this problem and solved it? I've seen posts in this tread from guys they use RPi & I2S with dam1021, so they should discover this, or just this 0.99 firmware has bug.
Maybe Soren can help?
Regards
Jacek
I think I've found solution :
I use in mpd.conf line: audio_output_format "*:*:2"
when I change this line to : audio_output_format "*:24:2" dam1021 starts play music sampled 44.1kHz - 24 bits should be fixed in mpd configuration.
Sorry Soren - it's not a bug in firmware 0.99 😉
Regards
Jacek
The 47uf is a MLCC X5R, the 10nF and 100pf are MLCCs C0G (NP0) with Flexible (Soft) termination, so they can take a little more soldering abuse 🙂
I did upsampling just to make dam1021 running with 44.1kHz and I2S.
Perhaps somebody will have the same issue and it's good to do full picture:
I did few different settings in mpd.conf for audio_output_format parameter.
Part of my mpd config looks like:
#******************************
audio_output {
type "alsa"
name "I2S"
device "hw:0,0"....
format "*:*:2"....
mixer_type "disable"
auto_resample "no"
auto_channels "no"
auto_format "yes"
}
audio_output_format "*:*:2"
#*********************************
command: cat /proc/asound/card0/pcm0p/sub0/hw_params shows:
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 4410
buffer_size: 22050
and dam1021 is not locked and not play.
when I start play music (using above config) with rate higher then 44100 DAC locks and play,
or
when I change parameter to audio_output_format "*:24:2" DAC also locks and play every sample rates from 44.1kHz to 192kHz.
So in my case dam1021 doesn't play 16bit/44.1 kHz from I2S input.
Right now I think it is not big problem because when audio_output_format is set to 24 bit DAC works ok and I guess mpd with this settings and 16/44.1 music just add 8 missing least significant bits without any resampling and distortion.... but any way this is a little bit odd.
Jacek
Hi Danny,
If Bernard is ok, it could be great you benchmark your moded Dam. and Bernard Aya 2 with my moded caped AYA 2 with :Ian FIFO 2, Clock 2+Laptech crystals 22.x+24.x / 8.x as alternative (have the 3), Simultaned mode I2S to PCM board + Wave I/O + all R-Core feeded ! Bruxelles Meeting ?
(PM if you are ok with Bernard. Of course french wine provided by me 🙂 )
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Hi Eldam,
That would be nice, I'll send you a PM.
Your AYA II is seriously pimped 🙂
We'll bring the Belgian beers
Regards,
Danny
That would be nice, I'll send you a PM.
Your AYA II is seriously pimped 🙂
We'll bring the Belgian beers
Regards,
Danny
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Power-on volume problem
Soeren,
If DAM1021 is powered on while music signal is present on the selected digital input, there is a rather annoying full volume sound blast before the dac sets its volume according to potentiometer position.
To make it easy to see, the attached image shows the problem while playing a sine wave. After power on, there is a 150ms full volume blast of undistorted sound before it does to normal volume as set by volume pot. Yellow channel is power, cyan is unbuffered output. It happens with all digital inputs.
It would be good to have it fixed in the next firmware release.
Serial port output during DAM1021 power-on sequence:
Soeren,
If DAM1021 is powered on while music signal is present on the selected digital input, there is a rather annoying full volume sound blast before the dac sets its volume according to potentiometer position.
To make it easy to see, the attached image shows the problem while playing a sine wave. After power on, there is a 150ms full volume blast of undistorted sound before it does to normal volume as set by volume pot. Yellow channel is power, cyan is unbuffered output. It happens with all digital inputs.
It would be good to have it fixed in the next firmware release.
Serial port output during DAM1021 power-on sequence:
Code:
R0.99
F5
V+00 <-- This is where the full volume blast starts
I0
V-30 <-- 150ms later volume goes to normal
L044
dam1021 uManager Rev 0.99 20150826 FPGA Rev 0.99 Press ? for help.
# set
Conspeed = 115200
Volume = -99
Filter = Mixed
Mode = NormalAttachments
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Hi, refer to past post, dam1021 req 64fs BCLK, you can check the output i2s signal .
the i2s output is 32fs acturally.I found this issue on raspberry pi 2 board with default i2s driver, I modified the ratio in source code to output 64fs BCLK.
I am interested in purchasing this project but have one concern. The single ended output of 1.4V is only 70% of the industry norm of 2V. Does it come across as a 'quiet' DAC with limited dynamics?
Thanks
Cyclopse
Thanks
Cyclopse
I have been using the hifiberry I2S driver, it produces 64fs signal.
In Raspbian, add these lines to /boot/config.txt:
Code:
dtparam=i2s=on
dtoverlay=hifiberry-dacHi, you can download source from office git server:
git clone --depth=1 https://github.com/raspberrypi/linux
@hifiberry_dac.c file, line31, you can see below codes:
static int snd_rpi_hifiberry_dac_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int sample_bits =
snd_pcm_format_physical_width(params_format(params));
return snd_soc_dai_set_bclk_ratio(cpu_dai, sample_bits * 2);
}
when the audio file format is 16bit, it will output 16*2=32 fs.
I guess you are using 24bit 44.1KHz format file, so you can get 64fs BCLK finally.
PS: I double check with logic analyzer, raspi 2 board's i2s do output 32bit BCLK when you play 16bit 44.1Khz audio file.
Best Regards!
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