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Budget Standalone "Toslink > DSP > Toslink" with Camilladsp. Set up instructions for newbies.

Ah okay, thought so.
btw found this upsampler, might be a better option then MCHStreamer. what do you think?

That is a hardware ASRC, if you had that between your source and your RPi it could convert all inputs to a common sample rate. You could then run CamillaDSP at a constant sample rate.

Michael
 
That is a hardware ASRC, if you had that between your source and your RPi it could convert all inputs to a common sample rate. You could then run CamillaDSP at a constant sample rate.

Michael
yes, and it accepts coaxial which is fine. And I guess I can hook it up via i2s to raspberry (although I have to dump my transport hat as there is only one i2s line and use something like usb to tosslink)
As a cheaper alternative DIR9001 + SRC4192I might work but I'm not a DIY expert.
Or even this thing which looks like it does everything all in one
The latter looks more appealing though
 
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:eek: this can be a better deal than the 1eur hdmi extractor. Will try to get one, thanks!
48/16 still perfectly fits TV usage
Any (audible) reason why 48/16 would not be enough for music?
 
Any (audible) reason why 48/16 would not be enough for music?
not that i know of, i believe camilladsp will still use a higher bit depth for processing -i might be wrong though-. I said perfect for TV usage because TVs typically send 48kHz/16 bit via toslink
 
not that i know of, i believe camilladsp will still use a higher bit depth for processing -i might be wrong though-. I said perfect for TV usage because TVs typically send 48kHz/16 bit via toslink
Cdsp appears to process at float64 internally indeed.
Regarding sample rate, I just found this post from Michael about 48 vs 96 kHz, probably inaudible, I mean I just trust the guy :) and the slight extra warping at high freqs means my own ears will be the limiting factor anyway.
 
Biggest issue to me is 16 bit output will have a lot of residual noise unless you have an analog volume control. Assuming you are using a DAC with 2 V output, residual noise at the DAC output will be 30 uV, which will then be multiplied by amplifier gain.

A big advantage of 24 or 32 bit output to the DAC is that at lower levels digital volume control will attenuate the noise from your source and push it below the noise floor of your DAC + amplifier.

For example, take a 16 bit source and 32 bit output to the DAC. If you have a DAC with 112 dB dynamic range at 2 V output, coupled with an amplifier that has 100 dB dynamic range at 5 W in to 4 ohm and 25.6 dB gain, at -20 dB digital volume control position residual noise at the speaker terminals is 121 uV which is pretty good.

If you only have 16 bit output to DAC with the same system residual noise at the speaker terminals will be 591 uV at digital all volume control levels. In this system a lower noise DAC or amplifier would gain you almost nothing.

Michael
 
Biggest issue to me is 16 bit output will have a lot of residual noise unless you have an analog volume control. Assuming you are using a DAC with 2 V output, residual noise at the DAC output will be 30 uV, which will then be multiplied by amplifier gain.

A big advantage of 24 or 32 bit output to the DAC is that at lower levels digital volume control will attenuate the noise from your source and push it below the noise floor of your DAC + amplifier.

For example, take a 16 bit source and 32 bit output to the DAC. If you have a DAC with 112 dB dynamic range at 2 V output, coupled with an amplifier that has 100 dB dynamic range at 5 W in to 4 ohm and 25.6 dB gain, at -20 dB digital volume control position residual noise at the speaker terminals is 121 uV which is pretty good.

If you only have 16 bit output to DAC with the same system residual noise at the speaker terminals will be 591 uV at digital all volume control levels. In this system a lower noise DAC or amplifier would gain you almost nothing.

Michael
Thank you very much for the detailed explanation Michael, that makes a lot of sense. As I do not have an analog volume control in my current setup, it is very relevant to me.

I have been looking for cheap-ish or second hand sound card that would do spdif output (for speakers through decent dac) + analog output (for subwoofer), in the 24/32 bit realm but I have not found much.

Only contendant so far is Sound Blaster Extigy at £32 used on ebay, lots of I/O and seems to work well in linux. Anyone tried that?
 
As an alternative I suppose I could use two of these usb -> spdif and a spare cheap dac for the sub.
I am assuming the spdif from both units will be clock synced?
 
Any (audible) reason why 48/16 would not be enough for music?
i use terratec aureon 7.1 usb . 16bit 48khz (Cmedia CM6206) . Spdif in and out, 8 channels , buffer for headphones.
with my setup , digital volume with camillaDSP , JBL studio 590 ( 92 db / W) and aiyama A07 amp i can hear very little noise at 10 cm from horn.
if i use another soundcard i have ( TerraTec Aureon XFire HD 8.0 USB, 24 bit with 105 snr) this is dead silent.
But i dont hear at 10cm...
so using best soundcard change nothing in real use. 16/44.1(48) is all that is needed.
 
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i use terratec aureon 7.1 usb . 16bit 48khz (Cmedia CM6206) . Spdif in and out, 8 channels , buffer for headphones.
with my setup , digital volume with camillaDSP , JBL studio 590 ( 92 db / W) and aiyama A07 amp i can hear very little noise at 10 cm from horn.
if i use another soundcard i have ( TerraTec Aureon XFire HD 8.0 USB, 24 bit with 105 snr) this is dead silent.
But i dont hear at 10cm...
so using best soundcard change nothing in real use. 16/44.1(48) is all that is needed.
Thanks for the feedback, much appreciated.
I hear you and that was my feeling as well based on other people IRL experience.

However, we are constantly chasing low noise kits across the chain with good dacs and amps so I also feel that it would be a shame to 'waste' it because I was a bit cheap.

It seems that I could get away with it by investing a tiny bit more on either a higher bit depth processing sound card or 2x usb-spdif converters and recycling a spare dac.
 
Has anyone tried this one?
Funnily enough I have one plugged into my laptop right now. Linux identifies it as ICUSBAUDIO7D

Only just got this to play with and haven't spent much time with it. So far have tried patching (with qpwgraph) SPDIF input to speaker out, trying to record with Audacity & arecord from the command line.

Optical SPDIF input works just fine when fed from a Chromecast Audio (44.1 48Khz).
However, SPDIF does not work when fed from a Sony TV (which insists on 48Khz - I have it set to PCM always and this works fine with an SMSL DAC).

Allegedly this USB audio interface does support 48Khz though, so if anyone has any clues as to how to get that to work would love to hear.

I see above someone recommending forcing Camilla to 48Khz but haven't even got that far - if I can't get anything else to record from the TV don't see the point. Or does Camilla do something funky with hardware settings that magically make 48Khz input work - and if so, what?

EDIT: My bad - the Chromecast Audio is 48Khz too! Got DAC inputs mixed up - the only 44.1Khz one it has is from an MPD server via USB.
 
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Or does Camilla do something funky with hardware settings that magically make 48Khz input work
No magic unfortunately. So if for example the arecord tool isn't able to record at 48 kHz, then CamillaDSP won't be able to either. What happens when you try with arecord? Do you have anything other than the TV that can output 48 kHz on SPDIF? Could be worth testing with another device, just to check if there is something strange with the SPDIF signal from the TV that the USB adapter doesn't like.
 
What happens when you try with arecord?
It runs, but output is just silence.
Do you have anything other than the TV that can output 48 kHz on SPDIF?
Sadly not. In fact yes! I made a mistake with the Chromecast Audio which is 48Khz too, so sample rate doesn't appear to be the problem.

FWIW, I have just been playing around with EasyEffects. Again, Chromecast input works fine, TV does not.

What is interesting with that, is that EasyEffects is a pipewire thing, and pipewire allegedly runs at 48Khz. In fact, I can cat /proc/asound/ICUSBAUDIO7D/pcm0c/sub0/hw_params while it's running which yields:

access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 512
buffer_size: 32768


Ie. the hardware says it is actually sampling at 48Khz, for both Chromecast & TV, but only the Chromecast (which is actually 44.1 48Khz too) actually works.

:confused:

Edit: So after looking again at the sample rates my DAC (SMSL DO100) says each source is, both Chromecast Audio & TV are 48Khz. I checked the SPDIF cables and the one from TV is good so that can't be the issue. So if it's not sample rate, what is the problem? How come my DAC has no issue with the TV's optical SPDIF but this USB adapter does?
 
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Ie. the hardware says it is actually sampling at 48Khz, for both Chromecast & TV, but only the Chromecast (which is actually 44.1 48Khz too) actually works.
Are both Chromecast and TV signals encoded in 16 bits?
 
Are both Chromecast and TV signals encoded in 16 bits?
I have no idea. The DO100 DAC supports up to 24 bit but sadly the display only shows the sample rate.
I discovered the Android TV box our ISP supplied also has an optical SPDIF output, so tested that. Works just fine. From this I would assume that it's nothing to do with Android O/S and some funky business Sony have added on top?
FWIW, the model is a KD-55XE9005
Connecting to the TV using adb and running dumpsys gives a whole load of stuff I mostly don't understand so went searching for clues... can't find any notable references to sample rates other than 48Khz/16 bit.
 
Connecting to the TV using adb and running dumpsys gives a whole load of stuff I mostly don't understand so went searching for clues... can't find any notable references to sample rates other than 48Khz/16 bit.
48kHz/16bit is standard Android TV indeed so must be something else...
 
48kHz/16bit is standard Android TV indeed so must be something else...
I tried a different optical cable with no success.

I also have a bi-directional coax<->optical converter which is normally in use to convert optical from TV into coax for the DAC (as the DAC's optical input is taken)... connecting TV(optical)->converter(optical)->ICUSBAUDIO7D(optical) unsurprisingly doesn't work either.

There must be some kind of subtle format incompatibility with the TV output but I can't find any concrete details on Sony TV SPDIF specs so really don't know.
 
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