I just received it and it does work well, the sound is smooth and detailed so I can say there is no problem with me. I use a meridian DAC and DSP active speakers.
Using the spdif from the maudio 2496PCI from the PC was not tolerable. Every time I switch the rooms lights on/off the sound would stop for a second. Now everything is tip top sounding and no drop off!
Any with the same usb/spdif convertor and care to comment?
Using the spdif from the maudio 2496PCI from the PC was not tolerable. Every time I switch the rooms lights on/off the sound would stop for a second. Now everything is tip top sounding and no drop off!
Any with the same usb/spdif convertor and care to comment?
I have used them both with Ubuntu. I sold the X2 but have my DX connected to Ubuntu by USB at this moment. System IDs the unit as "Audiotrak_DR.DAC2_DX".
I am using the switching PS and USB cable that came in the box. The USB cable has a noise filter at the "B" (DAC) end.
Hi joe_mansell,
thanks for the info, have you tried if it's possible to use these cards with Ubuntu also at 96/24?
Ciao
Andrea
Everything going to the DAC was 44kHz, couldn't find a solution. I don't know if this was gstreamer or pulseaudio or ALSA issue.
To know what is going on you have to disable pulseaudio (maybe this is not necessary if you are shure no other apps are using audio) and try something like:
# aplay -l
to see the right numbers for you interface (to replace x and y below)
# aplay -Dhw:x,y song.wav
with song.wav a 96KHz 24bit wav file
Ciao
Andrea
I have a teralink X2 on my ubuntu netbook...works out of the box, but cannot get it to output 24b 96k
Can i ask you which DAC do you have after the X2 and in which way is connected (coaxial, toslink or i2s)?
Ciao
Andrea
from the X2 it goes into a behringer SCR2496 and then to a Altmann DAC...
all optical.
but so far the behringer shows he receive 16 bits @ 44.1 Khz...
I am exploring ubuntu for the right player...so far I am pleased with the vlc player...
I need to find some time to do some more testing...
I have used a trends usb dac till now...was also very good.
Wanted to see if 24/96 native sound would be something...
kind regards,
Tony
Thanks Tony really kind
i don't know your linux audio skill but if you are using pulseaudio you have everything resampled at a fixed samplerate / bit depth, so if you want to try 96/24 you have to configure your player to talk directly to alsa device.
The more direct way is to use 'aplay' from the prompt of a terminal window first to know the name of the alsa device:
andrea@supermicro:~# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0
in this example if i would like to play throu "Intel ICH - IEC958" i have to use card 0 and device 4 so the right command is:
andrea@supermicro:~# aplay -D hw:0,4 file.wav
or (if the first give you error about format)
andrea@supermicro:~# aplay -D plughw:0,4 file.wav
with file.wav a 96/24 wav file
Ciao Andrea
this is whatI have:
mini@toon-minilaptop:~$ aplay -l
**** Lijst van PLAYBACK hardware-apparaten ****
kaart 0: Intel [HDA Intel], apparaat 0: STAC92xx Analog [STAC92xx Analog]
Sub-apparaten: 1/1
Sub-apparaat #0: subdevice #0
thanks for the tip!
Strange, i see only the sound card integrated with the motherboard.
Had you connected the teralink x2 before entering the 'aplay -l' command?
Ciao
Andrea
correct...X2 was not connected at that time...
saved your command and I will try it the coming days...
there is a long weekend ahead!
This is not 100 percent true.
In /etc/pulse/default.pa, you can configure default sample rates for each device, thusly.
add-autoload-sink output module-alsa-sink device=plughw:0,0 rate=48000 sink_name=output
But these rates can also be set on the fly. It may be that Gstreamer and other player back ends are not yet able to switch sample rates and bit depths to match files though.
I have had fairly good success with pulseuadio, sample rates and different output devices (hdmi audio, usb 96/24 converter). This may be useful to others trying the same:
1. Sample rate and word-depth can be set on a per-user basis, with a file .pulse/daemon.conf in your $HOME directory. Copy the daemon.conf from /etc/pulse and customize the one in your own directory.
2. Set the following according to your source and output device capabilities:
resample-method = speex-float-2 (or src-sinc-medium-quality if you have a fast CPU)
default-sample-format = s24le
default-sample-rate = 96000
3. If pulse is already running,
$ pulseaudio -k
will kill / restart pulse, using your new settings.
4. If source is 96/24, then playing with these settings should NOT need resampling. If playing CD or 48Khz audio, the audio will upsample with the resampler defined. Checking the CPU usage should confirm this.
Comment: on a fairly fast Dell laptop, only src-sinc-medium resampling is reliable (20% CPU usage). src-sinc-best stutters. Also my UHFnanoS dac (ESI Dr DAC Nano), seems to sound best using 48kHz or 96kz. 44.1kHz sounds harsh.
As manually editing daemon.conf is a hassle, I wrote a little python utility that allows me to change sample/word rate via the applet tray, as I would for network manager
After the pain of figuring out pulseaudio, I find it works rather well. I have an outstanding issue using Alsa (without pulse) on the USB dac... see my next post.
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