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CamillaDSP - a flexible linux IIR and FIR engine for crossovers, room correction etc.
CamillaDSP - a flexible linux IIR and FIR engine for crossovers, room correction etc.
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Old 26th March 2020, 07:19 AM   #281
HenrikEnquist is offline HenrikEnquist  Sweden
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It looks as if squeezelite can send audio data to stdout, could you try piping that to CamillaDSP? Then you wouldn't need any Loopback. It's probably goood to set "queuelimit" to 1 for that.
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Old 26th March 2020, 07:33 AM   #282
phofman is offline phofman  Czech Republic
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Quote:
Originally Posted by lykkedk View Post
So it seems that squeezelite is locking the samplerate somehow (my guess!)
See my post above, it explains why your setup does not work.

The loopback module creates two connected alsa devices, sharing a common memory buffer. That means no resampling, no sample format change, only one writing pointer and one reading pointer (it is a bit more complicated because each device can setup different buffer/period size).

The first side to open defines the params for both sides. The other side must use the same params if it wants to open its-side device. Change can happen only when both devices are released/closed.

Use the /proc/asound files, they provide key information for alsa troubleshooting.
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Old 26th March 2020, 07:42 AM   #283
phofman is offline phofman  Czech Republic
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Actually there is the module parameter pcm_notify and PCM control pcm_notify (available in amixer).

IIUC the module param should allow closing the capture side (with error) when params change on playback side. That would be easy to use in camilladsp - just re-opening the device with correct params.
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Old 26th March 2020, 07:46 AM   #284
lykkedk is offline lykkedk  Denmark
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Quote:
Originally Posted by HenrikEnquist View Post
It looks as if squeezelite can send audio data to stdout, could you try piping that to CamillaDSP? Then you wouldn't need any Loopback. It's probably goood to set "queuelimit" to 1 for that.
set "queuelimit" to 1
Where to set this? in camilladsp ?

I have to figure out howto grab the stout from squeezelite into camilladsp also.
I guess i will have to change th .yml file for that to happend.

Will be back

Jesper.
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Old 26th March 2020, 07:47 AM   #285
lykkedk is offline lykkedk  Denmark
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phofman, thanks again.

I will study this.

Jesper.
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Old 26th March 2020, 09:03 AM   #286
cube75 is offline cube75  Netherlands
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Goodmorning,

I wondered about this before. Does using pipe output make things easier / simpler ?
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Old 26th March 2020, 09:35 AM   #287
lykkedk is offline lykkedk  Denmark
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phofman

Quote:
The first side to open defines the params for both sides. The other side must use the same params if it wants to open its-side device. Change can happen only when both devices are released/closed.

Use the /proc/asound files, they provide key information for alsa troubleshooting.
I understand this now.

What i do is to close/reload the camilladsp through python, samplerate is set there at output.
That part is working.
The part where i need to stop/reload/close<>open the squeezelite is not working.
As you wrote i need to stop booth devices.

Will go thinking about this

Jesper.
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Old 26th March 2020, 10:43 AM   #288
phofman is offline phofman  Czech Republic
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Try the pcm_notify param of the snd-aloop module. IMO it will keep the playback side unlocked and faulting capture side if the params change.
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Old 26th March 2020, 11:33 AM   #289
HenrikEnquist is offline HenrikEnquist  Sweden
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Quote:
Originally Posted by phofman View Post
Try the pcm_notify param of the snd-aloop module. IMO it will keep the playback side unlocked and faulting capture side if the params change.
Good tip! That options seems quite useful. Breaking the capture should make CamillaDSP exit, so then you can just restart with the right config.
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Old 26th March 2020, 12:18 PM   #290
lykkedk is offline lykkedk  Denmark
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Henrik

Someone at slimserver forum (LMS) asked me this :
Quote:
f you want/need to do some development rather than have scripts or modded squeezelite, wouldn't it be better for camilla DSP to accept a WAV input (sample rate,width in header in line with data) rather than plain PCM so that from each data stream it can determine on the fly the audio format. It would then just be a drop into LMS conf file.
With more explanation :
Quote:
The WAV header is a fixed number (44?) of bytes at start of file or stream. All camilladsp would have to do is on startup read the WAV header, analyse it ( Page not found | Jawad's Blog ) and then use the values as if they are default or from command line / config file.
Audio DSP processing then starts at byte 45. Chunk length should be zero to indicate infinity - it is not required in a streaming environment.

You might even ask the developer to do it as a command line option (e.g. -w) ?
This
Quote:
It would then just be a drop into LMS conf file
is just to configure for them using LMS howto transcode (flac etc...)

I hope it's okay to ask you if this could be possible to do what he suggest?

Regarding the other suggestions phofman and you told here (stdout & snd-aloop pcm_notify, sry.. i really havent figured it out yet, and i think the suggestion from the other forum was important)

Jesper.
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