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Dummy to dummy multichannel DSP on the ultra cheap instructions

MarcosCh

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Hello guys,

I have been reading and asking here and there about how to set up a multichannel system with DSP on the ultra cheap. And even though many of you helped me a lot to understand the implications and requirements and answer all my stupid questions (special thanks for @mdsimon2 in between others), I could not find any detailed instructions for complete dummies on how to achieve what I wanted without having to buy an expensive and in some cases difficult to get multichannel dac.

The final purpose in my case is to build a pair LXmini leaving the doors open for a pair of subs and without having to invest too much in the electronics. That is, 6 channels with crossovers… $$$

Reading mdsimon2 excellent instructions for multichannel dsp with camilladsp was very inspiring. I wanted to do exactly the same but in moodeaudio (why? Because I already use moode and it already has camilladsp implemented, so, why not) and on the ultra cheap.

The parts were all there:

- Raspberry pi 4b running moodeaudio with camilladsp

- A cheap usb multichannel soundcard proben to work with camilladsp that you can easily find ultra cheap used (30 euros in my case): Asus Xonar U7 (not the mkii)

Total investment: 85 euros

As I need the pi 4 for other purposes, next week I will check if it works in the new zero 2w, that I already have doing room correction with moode/Camilla. This would bring the total investment to less than 50 euros.

Will not sound like an Okto dac? Maybe, I don’t think I will ever have the chance to find out…

So there we go with the instructions. Again, these are instructions from newbie to newbie, the only objective is to make it work. Feel free to point out things that I have done wrong or in a stupidly unnecessary way, I am sure there are many.

Instructions:

These instructions assume you already have a pi with moodeaudio running. There are many good instructions on how to get there so i won't cover that. Any pi would do except the zero. The pi zero can certainly run moode, but not camilladdp, and there is no way to change this. The new zero 2w on the other hand runs moode and camilla just fine.

In order to create a new configuration from zero, the best is to make a copy of the "flat" configuration that comes preloaded in moode and modify it. To do this, go to "m" > configure > camilladsp.

Scroll down to "Pipeline configuration" and in "pipeline" select "flat" from the dropdown menu.

pic 1.jpg


Once “flat” is selected, just below, click in "copy" and this creates a copy of the “flat” configuration to experiment with. Moode asks to give the new configuration a name. I used U7-8ch.

The configuration U7-8ch is created. Then, we need to modify this configuration to our needs:

We select the new configuration in the same page page: general > configuration > dropdown menu:

pic 2.jpg


Once selected, we need to enter the "pipeline configurator" to modify it:

Scroll down this same page and in pipeline editor, change status to "on" and Expert mode to "on" and then press in "open" pipeline editor to access the editor.

This gives you access to the standard camillagui. There are tabs for "devices" where you select from where the signal is taken and where the modified signal goes, "filters" where you can create different filters (you can create any number of filters you want to use, does not need to be one filter per channel, can be channels with no filter and channels with multiple filters). In "mixers" you assign what income channel goes to which outgoing channel (can be more than one). In "pipeline" you assign any filter you want to any channel you want. Finally, in "files" you can load a different configuration from the ones you have, save changes, etc.

At any stage in any of the tabs, to apply the changes you make to the current configuration, you can press "apply to DSP" at the left side of the screen, just below the name of the configuration you are working on:

pic 3.jpg


(In the picture above, we are working on the configuration U7-8ch but the configuration is not selected in moode at this moment, that’s why you dont see the VU meters with the sound and the state is "offline")


To create a working configuration for the U7:

Devices tab, bottom of the page:

Here we select the input and output devices and the number of channels in between other things.

I spent hours trying to use “loopback” as source for the sound as seen in many tutorials. Moode has the possibility to create a loopback just pressing one button, so it was very convenient. However, I must have done something wrong because I could not make it work.

In any case, I found that using the configuration for the capture device below, it works perfectly and camilladsp uses whatever you are streaming to moode as the input signal. Note that all the music I listen to is red book 16 bit 41 kHz, so the sample format selected (S16LE) should suffice. I have read somewhere that one can select a higher sample format than the actual music playing, but as this is not relevant for me, I left it like that.

The playback device is the U7. The name of this device in the system is hw:2,0. In this case I use plughw:2,0. I have read that plughw is safer than hw because it allows sample conversion if needed (?). Channels must be set to 8, that the U7 supports. Does not work if you put more or less than 8, must be 8. S16LE is again the sample rate that the card supports. I think the U7 supports higher rates as well, but i don’t need this for the time being.
*edit: see the suggestion of @mdsimon2 in post #2 to use hw:2,0 instead of plughw:2,0 for playback device. He knows much better than me and you (if you are following these instructions) what he is talking about, so my recommendation would be to do as he says. If it doesn't work, you can always switch back to plughw:2,0

pic 4.jpg


Once done, save the changes pressing “apply to DSP”. At this point you will get some error messages because the rest of the tabs are not adapted yet to the 8 channels. Don’t worry, will solve that soon.

FILTERS tab:

Here you create the filters, simply press the green "+" and select the type the filter you want, the frequency, the Q, etc. Create as many as you want and give each a name or not create any, as you need. I won’t get into this section for two reasons: you might need different filters than mine, and most importantly, I have no idea about creating filters.

In the example below I have created two filters just to play with. Pressing the green cross creates a new filter.

pic 5.jpg


MIXERS tab.

Here you need to assign what income channel goes to each of the outgoing 8 channels. You can route more than one income channel to the same outgoing channel, but you must assign something to each of the 8 outgoing channels, otherwise you will get a error message. (note for the ultradummies like me: channels are numbered starting with 0! So for instance the two incomming channels are 0 and 1, don’t waste your time wanting to find channels 1 and 2). Here you can also select the gain for each individual channel. Example (you cannot see the complete page in the screenshot, but I have assigned all the outgoing channels 0 to 7):

pic 6.jpg


PIPELINE tab:

Here you assign filters to channels. It is not mandatory to assign one filter to each channel. You can assign none or more than one. See below example.

pic 7.jpg


At the bottom of the screen, there is a button with a wave that shows you the signal flow:

pic 8.jpg


And basically that's it, when you are done you save the changes to the configuration pressing "apply to dsp" as discussed, exit the pipeline editor, select the configuration in camilladsp configuration, and when you play the music, it goes through the pipeline you have just designed.

When at work, if you go to pipeline configuration again, you will see tha VU meters for all the income and outcome filters at work:

pic 9.jpg


The channel assignment for the U7 is the following:

channel 0 : headphone L - RCA L

channel 1 : headphone R - RCA R

channel 2 : Ctr L

channel 3 : Ctr R

channel 4 : Rear L

channel 5 : Rear R

channel 6 : Side L

channel 7 : Sode R

And that’s it, there it is your 85 euros streamer/dsp/multichannel dac up and running (hopefully). I hope it helps folks out there and does not generate even more confusion to the poor newbies that only want to listen to their favourite music.

Next step, a decent 6 channel amp or board on the cheap! hahaha

Enjoy!
 
Last edited:

mdsimon2

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Very nice, this an awesome value!

I am not very familiar with Moode's implementation of CamillaDSP but with resampling enabled what happens when the input file sampling frequency changes? For example from 44.1 kHz to 48 kHz? Does the capture rate change and it resamples to 44.1 kHz per your configuration?

I would avoid using plughw for your DAC as you really do not want ALSA resampling anything. How / when it resamples largely depends on the answers to my questions above. My intuition says you will be better off using hw instead of plughw and disable resampling, the Moode default is to load a new configuration when the sample rate changes.

How cheap is a cheap on a 6 channel amp? I might have some ideas.

Michael
 
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MarcosCh

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Thanks for your note Michael. I could not experiment much with sample rates because all my music is in red book format, but i hear what you are saying and will add an edit.

Regarding the amp, i really don't know yet.... The U7 outputs only 1 volt but i was assuming (maybe wrong) that having 4 channels of amplification for two speakers would give you the same max volume than a regular 2 channel systems with a dac outputting 2 volts? In any case, and not knowing how much gain the eq for the lxmini "eats", i would go for something with not too low gain. I would also prefer to stay small and tidy. Powerwise i think 30-40 wats per channel would suffice. I don't need 6 channels at this moment, 4 would suffice, i can always buy a second 2 channel amp if i ever decide to go for the subs.
Pricewise my reference would be the price of two aiyimas a07, that is 170eur where i live. I don't like the idea of the aiyimas because of having 2 psus, 2 cables etc, but i think it is a fair price point reference.
 

mdsimon2

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If everything is at 44.1 kHz then none of what I brought up really matters :). You might consider resampling to 96 kHz in CamillaDSP to minimize high frequency warping but it is not the end of the world if you do not. The original LXmini was used a miniDSP 2x4 which runs at 48 kHz so lower sampling rates are fine.

4 channels of amplification in an active speaker really doesn't help you from a sensitivity perspective vs a passive speaker. Are you doing volume control in CamillaDSP (or somewhere else upstream)? If so you won't need to worry as much about the DSP causing clipping but CamillaDSP does have the nice feature of telling you when clipping occurs so you can attenuate more if needed. I do volume control downstream in my LXmini setup so I have some permanent attenuation set to make sure I never clip the DSP, but I also have a good amount of extra gain in my amplifiers so that I can compensate for lower level recordings.

I think going for something with higher gain probably makes sense. LXminis are not very sensitive so hiss shouldn't be much of an issue and it will give some more flexibility with the 1 V output. AIYIMAs are hard to beat for that price and they have quite a bit of gain. Recently I was looking at some of the ICEpower modules and the 100AS2/200AS2 modules which are also TI TP325x based are very cheap (68/78 USD) but you need to buy 20 of them to get that pricing and as individual modules they are a lot more expensive. The other issue is that they only have 24 dB of gain so not a viable option for your use case.

Michael
 
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MarcosCh

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A few more comments on the Xonar U7 and the sound blaster X3 as potential alternative.
The U7 has actually two dac chips, CS4398 for the main (RCA) outputs and headphone, CS4362 for the rest of the channels.
There are measurements of the U7 (not mkii) available online in diyaudio and head-fi and it seems that it suffers from degradation of its performance with certain windows drivers, similar to what was discussed after the measurements Amir did on the newer model U7 mkii. I am assuming that these issues are not relevant when used in Linux...

The X3 (another low cost, but not ultra low cost in my book though) uses one multichannel AK4458vn chip for all the 8 channels. Seems that this chip was used in some AVRs not so long ago. I don’t know if this is an improvement over the U7 or not really, but I leave it there.
I could not find info about the output voltage but someone that owns one told me it is ca. 2 volts. The X3 can still be found new for ca 130eur and being a newer model I struggle to find it in the used market for less than 80-100 eur.
I cannot tell if the X3 would work with camilladsp, and it is definitely bulkier than the U7, but I think it would make a very attractive alternative, especially if the price falls in the near future.
 
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MarcosCh

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If everything is at 44.1 kHz then none of what I brought up really matters :). You might consider resampling to 96 kHz in CamillaDSP to minimize high frequency warping but it is not the end of the world if you do not. The original LXmini was used a miniDSP 2x4 which runs at 48 kHz so lower sampling rates are fine.

4 channels of amplification in an active speaker really doesn't help you from a sensitivity perspective vs a passive speaker. Are you doing volume control in CamillaDSP (or somewhere else upstream)? If so you won't need to worry as much about the DSP causing clipping but CamillaDSP does have the nice feature of telling you when clipping occurs so you can attenuate more if needed. I do volume control downstream in my LXmini setup so I have some permanent attenuation set to make sure I never clip the DSP, but I also have a good amount of extra gain in my amplifiers so that I can compensate for lower level recordings.

I think going for something with higher gain probably makes sense. LXminis are not very sensitive so hiss shouldn't be much of an issue and it will give some more flexibility with the 1 V output. AIYIMAs are hard to beat for that price and they have quite a bit of gain. Recently I was looking at some of the ICEpower modules and the 100AS2/200AS2 modules which are also TI TP325x based are very cheap (68/78 USD) but you need to buy 20 of them to get that pricing and as individual modules they are a lot more expensive. The other issue is that they only have 24 dB of gain so not a viable option for your use case.

Michael
Thanks for the suggestions, will look at the resampling, to be honest, I had not think about it.
For the volume control, I normally do it in moode, but I cannot tell if it is done upstream or downstream of camilla, I will check this out (just looking at the green bars)
Yeah, for the amp, I really want to avoid stacking aiyimas 07, I really want a single board/little amp. I see that aiyima themselves sell multichannel tpa3255 boards, but can't find any info about their quality or reliability.... :-/
 

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One other thing before I forget, you definitely want to set your playback bit depth as high as you can (preferably 24 or 32 bits). As you have it set now you are basically imposing the 16 bit truncation mentioned in Amir's review. You might say that this shouldn't matter because you are only playing 16 bit files but it does because you do not play those files at full volume and digital volume control is destructive.

Let's use the U7 mkII as an example. From Amir's review it looks like the U7 mkII has a dynamic range of 106 dB at an output voltage of 1.14 V (had to pick this off the SINAD measurement so it is not exact) when playing 24 bit data. This implies a residual noise level of 1.14 x 10^(-106/20) = 5.7 uV which is really quite impressive. If you limit to 16 bit the dynamic range now becomes 96 dB and the residual noise becomes 1.14 x 10^(-96/20) = 18.1 uV which is pretty mediocre.

Now let's say you reduce the volume by -20 dB digitally. CamillaDSP is using 64 bit floats so it can do this operation with essentially no loss, the original -96 dBFS noise floor from your 16 bit file shifts down to -116 dBFS, so clearly the DAC noise floor is higher than this and will govern the resolution.

Because we have dropped our level by -20 dB our max output voltage is now 1.14 x 10^(-20/20) = 0.114 V. At 24 bit our effective dynamic range is now 20 x log (0.114 x 10^6 / 5.7) = 86 dB, so 10 dB off 16 bit resolution. At 16 bit our effective dynamic range is now 20 x log (0.114 x 10^6 / 18.1) = 76 dB, now 20 dB off 16 bit resolution.

All of this only get's worse once you multiply the DAC residual noise with an amplifier.

Michael
 
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MarcosCh

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Thank you. Actually I set 16 out of complete laziness because I understand that the setting cannot be higher than the capability of the day (correct me if I am wrong) and I wasn't sure if the dac was 16 or 24 bit capable. I read now it is 24 bit. Would this be the max I can set in camilla or it doesn't really matter?
 

mdsimon2

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Can you access terminal in Moode? I think I read an earlier post where there is some web SSH terminal functionality?

If so run aplay -l which should report your audio device names. Once you have a device name run cat /proc/asound/YOURDEVICENAME/stream0 and it should report the available sampling rates and bit depths.

Michael
 
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MarcosCh

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[email protected]:~ $ cat /proc/asound/U7/stream0
ASUS Xonar U7 at usb-0000:01:00.0-1.4, full speed : USB Audio

Playback:
Status: Stop
Interface 1
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000
Bits: 16

Capture:
Status: Stop
Interface 3
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 8 IN (ASYNC)
Rates: 44100, 48000
Bits: 16
 

mdsimon2

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Hmm, that is too bad, looks like you are limited to 16 bits. Not the end of the world, if you are using an AIYIMA A07 that probably comes close to governing from a noise perspective anyways.

Michael
 
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MarcosCh

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30 euros! :D
BTW, to your previous question, the volume takes effect before camilladsp
 
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MarcosCh

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Sorry @mdsimon2 , I must rectify. I had accidentally plugged the card in one of the black USB sockets. Plugging it in one of the blue ones I get the following (I knew I had seen this before!). I assume I can change to S24_3LE as you suggested?


[email protected]:~ $ cat /proc/asound/U7/stream0
ASUS Xonar U7 at usb-0000:01:00.0-1.2, high speed : USB Audio

Playback:
Status: Stop
Interface 1
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 3
Format: S16_LE
Channels: 4
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 4
Format: S24_3LE
Channels: 4
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 5
Format: S16_LE
Channels: 6
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 6
Format: S24_3LE
Channels: 6
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 7
Format: S16_LE
Channels: 8
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 8
Format: S24_3LE
Channels: 8
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24

Capture:
Status: Stop
Interface 3
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 8 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 3
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 8 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
 
Last edited:

dc655321

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Can you access terminal in Moode? I think I read an earlier post where there is some web SSH terminal functionality?

If so run aplay -l which should report your audio device names. Once you have a device name run cat /proc/asound/YOURDEVICENAME/stream0 and it should report the available sampling rates and bit depths.

Michael


Could also look at supported audio params and their ranges via:

Code:
┌─ [[email protected]]:[~/Downloads]
└─>  aplay -D hw:0,0 --dump-hw-params /dev/zero
Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono
HW Params of device "hw:0,0":
--------------------
ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT:  S16_LE S32_LE
SUBFORMAT:  STD
SAMPLE_BITS: [16 32]
FRAME_BITS: [32 64]
CHANNELS: 2
RATE: 48000
PERIOD_TIME: (333 170667)
PERIOD_SIZE: [16 8192]
PERIOD_BYTES: [128 65536]
PERIODS: [2 32]
BUFFER_TIME: (666 341334)
BUFFER_SIZE: [32 16384]
BUFFER_BYTES: [128 65536]
TICK_TIME: ALL
--------------------
aplay: set_params:1374: Channels count non available
 
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MarcosCh

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Yup, thanks, seems to bring the same result:

[email protected]:~ $ aplay -D hw:2,0 --dump-hw-params /dev/zero
Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono
HW Params of device "hw:2,0":
--------------------
ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT: S16_LE S24_3LE
SUBFORMAT: STD
SAMPLE_BITS: [16 24]
FRAME_BITS: [32 192]
CHANNELS: [2 8]
RATE: [44100 192000]
PERIOD_TIME: [125 2972155)
PERIOD_SIZE: [6 131072]
PERIOD_BYTES: [64 524288]
PERIODS: [2 1024]
BUFFER_TIME: (62 5944309)
BUFFER_SIZE: [12 262144]
BUFFER_BYTES: [64 1048576]
TICK_TIME: ALL
--------------------
aplay: set_params:1339: Sample format non available
Available formats:
- S16_LE
- S24_3LE
 

mdsimon2

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Sorry @mdsimon2 , I must rectify. I had accidentally plugged the card in one of the black USB sockets. Plugging it in one of the blue ones I get the following (I knew I had seen this before!). I assume I can change to S24_3LE as you suggested?


[email protected]:~ $ cat /proc/asound/U7/stream0
ASUS Xonar U7 at usb-0000:01:00.0-1.2, high speed : USB Audio

Playback:
Status: Stop
Interface 1
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 3
Format: S16_LE
Channels: 4
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 4
Format: S24_3LE
Channels: 4
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 5
Format: S16_LE
Channels: 6
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 6
Format: S24_3LE
Channels: 6
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24
Interface 1
Altset 7
Format: S16_LE
Channels: 8
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 1
Altset 8
Format: S24_3LE
Channels: 8
Endpoint: 5 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24

Capture:
Status: Stop
Interface 3
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 8 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 16
Interface 3
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 8 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Bits: 24

Yes, but it needs to be in the right format for CamillaDSP so S24LE3. I would also get rid of the plughw while you are at it and see if everything still works.

Michael
 
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MarcosCh

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Done, all working :)
Will modify the instructions "make sure you plug the card in a blue usb socket" dummies like me need this sort of information...
20220207_223115.jpg
 

AbacusLX

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I did it! Thank you.
For an LXmini+2 with the U7 at first.
Then with a Sound Blaster X3, works perfectly. Advantage: format up to S32_LE and 8 identical DAC outputs (if LX521 one day).
But as I prefer LMS to Moode, I switched to the mdsimon2 version.
NB : There is a b-stock on Creative website, with 43% discount.
 
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MarcosCh

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Glad to hear you managed with the X3. I have a couple of soundblasters at home (not the X3 though) and they are a nightmare to set up in alsamixer.
Was also considering the X3 beacuse of 2V output vs the 1V output of the u7 but my experience with the other soundblasters prevented me from getting it.
Thanks for reporting that it works :)
 
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